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711 lines
21 KiB
711 lines
21 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 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 libavcodec/flacdec.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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#define ALT_BITSTREAM_READER |
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#include "libavutil/crc.h" |
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#include "avcodec.h" |
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#include "bitstream.h" |
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#include "golomb.h" |
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#include "flac.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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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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|
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typedef struct FLACContext { |
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FLACSTREAMINFO |
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AVCodecContext *avctx; ///< parent AVCodecContext |
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GetBitContext gb; ///< GetBitContext initialized to start at the current frame |
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int blocksize; ///< number of samples in the current frame |
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int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits |
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int sample_shift; ///< shift required to make output samples 16-bit or 32-bit |
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int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit |
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enum decorrelation_type decorrelation; ///< channel decorrelation type in the current frame |
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int32_t *decoded[MAX_CHANNELS]; ///< decoded samples |
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uint8_t *bitstream; |
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unsigned int bitstream_size; |
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unsigned int bitstream_index; |
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unsigned int allocated_bitstream_size; |
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} FLACContext; |
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|
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static const int sample_rate_table[] = |
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{ 0, |
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88200, 176400, 192000, |
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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 const int sample_size_table[] = |
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{ 0, 8, 12, 0, 16, 20, 24, 0 }; |
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|
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static const 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 int64_t get_utf8(GetBitContext *gb) |
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{ |
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int64_t val; |
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GET_UTF8(val, get_bits(gb, 8), return -1;) |
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return val; |
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} |
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static void allocate_buffers(FLACContext *s); |
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static int metadata_parse(FLACContext *s); |
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|
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static av_cold 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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avctx->sample_fmt = SAMPLE_FMT_S16; |
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if (avctx->extradata_size > 4) { |
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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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ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, |
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avctx->extradata); |
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allocate_buffers(s); |
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} else { |
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init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8); |
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metadata_parse(s); |
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} |
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} |
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return 0; |
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} |
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static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) |
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{ |
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av_log(avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d\n", s->min_blocksize, |
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s->max_blocksize); |
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av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize); |
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av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate); |
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av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels); |
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av_log(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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{ |
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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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// FIXME header overhead |
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s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; |
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} |
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for (i = 0; i < s->channels; i++) { |
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s->decoded[i] = av_realloc(s->decoded[i], |
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sizeof(int32_t)*s->max_blocksize); |
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} |
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if (s->allocated_bitstream_size < s->max_framesize) |
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s->bitstream= av_fast_realloc(s->bitstream, |
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&s->allocated_bitstream_size, |
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s->max_framesize); |
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} |
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void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, |
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const uint8_t *buffer) |
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{ |
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GetBitContext gb; |
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init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8); |
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/* mandatory streaminfo */ |
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s->min_blocksize = get_bits(&gb, 16); |
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s->max_blocksize = get_bits(&gb, 16); |
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skip_bits(&gb, 24); /* skip min frame size */ |
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s->max_framesize = get_bits_long(&gb, 24); |
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s->samplerate = get_bits_long(&gb, 20); |
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s->channels = get_bits(&gb, 3) + 1; |
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s->bps = get_bits(&gb, 5) + 1; |
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avctx->channels = s->channels; |
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avctx->sample_rate = s->samplerate; |
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avctx->bits_per_raw_sample = s->bps; |
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if (s->bps > 16) |
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avctx->sample_fmt = SAMPLE_FMT_S32; |
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else |
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avctx->sample_fmt = SAMPLE_FMT_S16; |
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s->samples = get_bits_long(&gb, 32) << 4; |
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s->samples |= get_bits_long(&gb, 4); |
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skip_bits(&gb, 64); /* md5 sum */ |
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skip_bits(&gb, 64); /* md5 sum */ |
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dump_headers(avctx, s); |
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} |
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/** |
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* Parse a list of metadata blocks. This list of blocks must begin with |
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* the fLaC marker. |
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* @param s the flac decoding context containing the gb bit reader used to |
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* parse metadata |
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* @return 1 if some metadata was read, 0 if no fLaC marker was found |
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*/ |
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static int metadata_parse(FLACContext *s) |
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{ |
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int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0; |
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int initial_pos= get_bits_count(&s->gb); |
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if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) { |
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skip_bits(&s->gb, 32); |
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do { |
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metadata_last = get_bits1(&s->gb); |
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metadata_type = get_bits(&s->gb, 7); |
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metadata_size = get_bits_long(&s->gb, 24); |
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if (get_bits_count(&s->gb) + 8*metadata_size > s->gb.size_in_bits) { |
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skip_bits_long(&s->gb, initial_pos - get_bits_count(&s->gb)); |
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break; |
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} |
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if (metadata_size) { |
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switch (metadata_type) { |
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case FLAC_METADATA_TYPE_STREAMINFO: |
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ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, |
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s->gb.buffer+get_bits_count(&s->gb)/8); |
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streaminfo_updated = 1; |
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default: |
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for (i = 0; i < metadata_size; i++) |
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skip_bits(&s->gb, 8); |
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} |
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} |
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} while (!metadata_last); |
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if (streaminfo_updated) |
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allocate_buffers(s); |
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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_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 > 1) { |
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av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n", |
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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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if (pred_order > samples) { |
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av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", |
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pred_order, samples); |
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return -1; |
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} |
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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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tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5); |
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if (tmp == (method_type == 0 ? 15 : 31)) { |
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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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} else { |
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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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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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const int blocksize = s->blocksize; |
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int32_t *decoded = s->decoded[channel]; |
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int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i; |
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/* warm up samples */ |
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for (i = 0; i < pred_order; i++) { |
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decoded[i] = get_sbits(&s->gb, s->curr_bps); |
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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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if (pred_order > 0) |
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a = decoded[pred_order-1]; |
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if (pred_order > 1) |
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b = a - decoded[pred_order-2]; |
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if (pred_order > 2) |
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c = b - decoded[pred_order-2] + decoded[pred_order-3]; |
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if (pred_order > 3) |
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d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4]; |
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switch (pred_order) { |
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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 < blocksize; i++) |
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decoded[i] = a += decoded[i]; |
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break; |
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case 2: |
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for (i = pred_order; i < blocksize; i++) |
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decoded[i] = a += b += decoded[i]; |
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break; |
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case 3: |
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for (i = pred_order; i < blocksize; i++) |
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decoded[i] = a += b += c += decoded[i]; |
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break; |
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case 4: |
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for (i = pred_order; i < blocksize; i++) |
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decoded[i] = a += b += c += d += decoded[i]; |
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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 i, j; |
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int coeff_prec, qlevel; |
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int coeffs[pred_order]; |
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int32_t *decoded = s->decoded[channel]; |
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/* warm up samples */ |
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for (i = 0; i < pred_order; i++) { |
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decoded[i] = get_sbits(&s->gb, s->curr_bps); |
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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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av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n"); |
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return -1; |
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} |
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qlevel = get_sbits(&s->gb, 5); |
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if (qlevel < 0) { |
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av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n", |
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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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coeffs[i] = get_sbits(&s->gb, coeff_prec); |
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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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if (s->bps > 16) { |
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int64_t sum; |
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for (i = pred_order; i < s->blocksize; i++) { |
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sum = 0; |
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for (j = 0; j < pred_order; j++) |
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sum += (int64_t)coeffs[j] * decoded[i-j-1]; |
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decoded[i] += sum >> qlevel; |
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} |
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} else { |
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for (i = pred_order; i < s->blocksize-1; i += 2) { |
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int c; |
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int d = decoded[i-pred_order]; |
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int s0 = 0, s1 = 0; |
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for (j = pred_order-1; j > 0; j--) { |
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c = coeffs[j]; |
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s0 += c*d; |
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d = decoded[i-j]; |
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s1 += c*d; |
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} |
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c = coeffs[0]; |
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s0 += c*d; |
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d = decoded[i] += s0 >> qlevel; |
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s1 += c*d; |
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decoded[i+1] += s1 >> qlevel; |
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} |
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if (i < s->blocksize) { |
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int sum = 0; |
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for (j = 0; j < pred_order; j++) |
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sum += coeffs[j] * decoded[i-j-1]; |
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decoded[i] += sum >> qlevel; |
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} |
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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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av_log(s->avctx, AV_LOG_ERROR, "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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|
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if (get_bits1(&s->gb)) { |
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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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} |
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//FIXME use av_log2 for types |
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if (type == 0) { |
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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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} else if (type == 1) { |
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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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} else if ((type >= 8) && (type <= 12)) { |
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if (decode_subframe_fixed(s, channel, type & ~0x8) < 0) |
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return -1; |
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} else if (type >= 32) { |
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if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0) |
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return -1; |
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} else { |
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av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n"); |
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return -1; |
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} |
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if (wasted) { |
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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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return 0; |
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} |
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|
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static int decode_frame(FLACContext *s, int alloc_data_size) |
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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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blocksize_code = get_bits(&s->gb, 4); |
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sample_rate_code = get_bits(&s->gb, 4); |
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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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av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", |
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assignment, s->channels); |
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return -1; |
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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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av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", |
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sample_size_code); |
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return -1; |
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} |
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if (bps > 16) { |
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s->avctx->sample_fmt = SAMPLE_FMT_S32; |
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s->sample_shift = 32 - bps; |
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s->is32 = 1; |
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} else { |
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s->avctx->sample_fmt = SAMPLE_FMT_S16; |
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s->sample_shift = 16 - bps; |
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s->is32 = 0; |
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} |
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s->bps = s->avctx->bits_per_raw_sample = bps; |
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|
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if (get_bits1(&s->gb)) { |
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av_log(s->avctx, AV_LOG_ERROR, "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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|
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if (blocksize_code == 0) |
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blocksize = s->min_blocksize; |
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else if (blocksize_code == 6) |
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blocksize = get_bits(&s->gb, 8)+1; |
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else if (blocksize_code == 7) |
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blocksize = get_bits(&s->gb, 16)+1; |
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else |
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blocksize = blocksize_table[blocksize_code]; |
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|
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if (blocksize > s->max_blocksize) { |
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av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize, |
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s->max_blocksize); |
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return -1; |
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} |
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|
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if (blocksize * s->channels * sizeof(int16_t) > alloc_data_size) |
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return -1; |
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|
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if (sample_rate_code == 0) |
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samplerate= s->samplerate; |
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else if (sample_rate_code < 12) |
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samplerate = sample_rate_table[sample_rate_code]; |
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else if (sample_rate_code == 12) |
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samplerate = get_bits(&s->gb, 8) * 1000; |
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else if (sample_rate_code == 13) |
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samplerate = get_bits(&s->gb, 16); |
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else if (sample_rate_code == 14) |
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samplerate = get_bits(&s->gb, 16) * 10; |
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else { |
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av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n", |
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sample_rate_code); |
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return -1; |
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} |
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|
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skip_bits(&s->gb, 8); |
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crc8 = av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, |
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s->gb.buffer, get_bits_count(&s->gb)/8); |
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if (crc8) { |
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av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8); |
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return -1; |
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} |
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|
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s->blocksize = blocksize; |
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s->samplerate = samplerate; |
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s->bps = bps; |
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s->decorrelation= decorrelation; |
|
|
|
// dump_headers(s->avctx, (FLACStreaminfo *)s); |
|
|
|
/* subframes */ |
|
for (i = 0; i < s->channels; 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, |
|
const uint8_t *buf, int buf_size) |
|
{ |
|
FLACContext *s = avctx->priv_data; |
|
int tmp = 0, i, j = 0, input_buf_size = 0; |
|
int16_t *samples_16 = data; |
|
int32_t *samples_32 = data; |
|
int alloc_data_size= *data_size; |
|
|
|
*data_size=0; |
|
|
|
if (s->max_framesize == 0) { |
|
s->max_framesize= FFMAX(4, buf_size); // 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 |
|
if (s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C')) |
|
buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize)); |
|
input_buf_size= buf_size; |
|
|
|
if (s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index) |
|
return -1; |
|
|
|
if (s->allocated_bitstream_size < s->bitstream_size + buf_size) |
|
s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_size + buf_size); |
|
|
|
if (s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size) { |
|
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 && input_buf_size) { |
|
return input_buf_size; |
|
} |
|
} |
|
|
|
init_get_bits(&s->gb, buf, buf_size*8); |
|
|
|
if (metadata_parse(s)) |
|
goto end; |
|
|
|
tmp = show_bits(&s->gb, 16); |
|
if ((tmp & 0xFFFE) != 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) & 0xFFFE) != 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, alloc_data_size) < 0) { |
|
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n"); |
|
s->bitstream_size=0; |
|
s->bitstream_index=0; |
|
return -1; |
|
} |
|
|
|
#define DECORRELATE(left, right)\ |
|
assert(s->channels == 2);\ |
|
for (i = 0; i < s->blocksize; i++) {\ |
|
int a= s->decoded[0][i];\ |
|
int b= s->decoded[1][i];\ |
|
if (s->is32) {\ |
|
*samples_32++ = (left) << s->sample_shift;\ |
|
*samples_32++ = (right) << s->sample_shift;\ |
|
} else {\ |
|
*samples_16++ = (left) << s->sample_shift;\ |
|
*samples_16++ = (right) << s->sample_shift;\ |
|
}\ |
|
}\ |
|
break; |
|
|
|
switch (s->decorrelation) { |
|
case INDEPENDENT: |
|
for (j = 0; j < s->blocksize; j++) { |
|
for (i = 0; i < s->channels; i++) { |
|
if (s->is32) |
|
*samples_32++ = s->decoded[i][j] << s->sample_shift; |
|
else |
|
*samples_16++ = s->decoded[i][j] << s->sample_shift; |
|
} |
|
} |
|
break; |
|
case LEFT_SIDE: |
|
DECORRELATE(a,a-b) |
|
case RIGHT_SIDE: |
|
DECORRELATE(a+b,b) |
|
case MID_SIDE: |
|
DECORRELATE( (a-=b>>1) + b, a) |
|
} |
|
|
|
*data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2); |
|
|
|
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 av_cold 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, |
|
CODEC_CAP_DELAY, |
|
.flush= flac_flush, |
|
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"), |
|
};
|
|
|