X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fflac.c;h=07da702e9495fc4f5afdc7f032e75ff8e45ed446;hb=e00eb03cd8bfb6993d33c30ccd560947b5f6bad5;hp=8bf00b2d047ce5c26788fdaae6f44f28b847b867;hpb=0187178e07218772553767da0cef12e0c0b149a6;p=ffmpeg diff --git a/libavcodec/flac.c b/libavcodec/flac.c index 8bf00b2d047..07da702e949 100644 --- a/libavcodec/flac.c +++ b/libavcodec/flac.c @@ -1,804 +1,245 @@ /* - * FLAC (Free Lossless Audio Codec) decoder - * Copyright (c) 2003 Alex Beregszaszi + * FLAC common code + * Copyright (c) 2009 Justin Ruggles * - * This library is free software; you can redistribute it and/or + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. + * version 2.1 of the License, or (at your option) any later version. * - * This library is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -/** - * @file flac.c - * FLAC (Free Lossless Audio Codec) decoder - * @author Alex Beregszaszi - * - * For more information on the FLAC format, visit: - * http://flac.sourceforge.net/ - * - * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed - * through, starting from the initial 'fLaC' signature; or by passing the - * 34-byte streaminfo structure through avctx->extradata[_size] followed - * by data starting with the 0xFFF8 marker. - */ - -#include - -#include "avcodec.h" -#include "bitstream.h" -#include "golomb.h" -#include "crc.h" - -#undef NDEBUG -#include - -#define MAX_CHANNELS 8 -#define MAX_BLOCKSIZE 65535 -#define FLAC_STREAMINFO_SIZE 34 - -enum decorrelation_type { - INDEPENDENT, - LEFT_SIDE, - RIGHT_SIDE, - MID_SIDE, -}; - -typedef struct FLACContext { - AVCodecContext *avctx; - GetBitContext gb; - - int min_blocksize, max_blocksize; - int min_framesize, max_framesize; - int samplerate, channels; - int blocksize/*, last_blocksize*/; - int bps, curr_bps; - enum decorrelation_type decorrelation; - - int32_t *decoded[MAX_CHANNELS]; - uint8_t *bitstream; - int bitstream_size; - int bitstream_index; - unsigned int allocated_bitstream_size; -} FLACContext; - -#define METADATA_TYPE_STREAMINFO 0 - -static int sample_rate_table[] = -{ 0, 0, 0, 0, - 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000, - 0, 0, 0, 0 }; - -static int sample_size_table[] = -{ 0, 8, 12, 0, 16, 20, 24, 0 }; - -static int blocksize_table[] = { - 0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0, -256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7 +#include "libavutil/audioconvert.h" +#include "libavutil/crc.h" +#include "libavutil/log.h" +#include "bytestream.h" +#include "get_bits.h" +#include "flac.h" +#include "flacdata.h" + +static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 }; + +static const int64_t flac_channel_layouts[6] = { + AV_CH_LAYOUT_MONO, + AV_CH_LAYOUT_STEREO, + AV_CH_LAYOUT_SURROUND, + AV_CH_LAYOUT_QUAD, + AV_CH_LAYOUT_5POINT0, + AV_CH_LAYOUT_5POINT1 }; static int64_t get_utf8(GetBitContext *gb) { - uint64_t val; - int ones=0, bytes; - - while(get_bits1(gb)) - ones++; - - if (ones==0) bytes=0; - else if(ones==1) return -1; - else bytes= ones - 1; - - val= get_bits(gb, 7-ones); - while(bytes--){ - const int tmp = get_bits(gb, 8); - - if((tmp>>6) != 2) - return -1; - val<<=6; - val|= tmp&0x3F; - } + int64_t val; + GET_UTF8(val, get_bits(gb, 8), return -1;) return val; } -#if 0 -static int skip_utf8(GetBitContext *gb) +int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, + FLACFrameInfo *fi, int log_level_offset) { - int ones=0, bytes; + int bs_code, sr_code, bps_code; - while(get_bits1(gb)) - ones++; - - if (ones==0) bytes=0; - else if(ones==1) return -1; - else bytes= ones - 1; - - skip_bits(gb, 7-ones); - while(bytes--){ - const int tmp = get_bits(gb, 8); - - if((tmp>>6) != 2) - return -1; - } - return 0; -} -#endif - -static void metadata_streaminfo(FLACContext *s); -static void dump_headers(FLACContext *s); - -static int flac_decode_init(AVCodecContext * avctx) -{ - FLACContext *s = avctx->priv_data; - s->avctx = avctx; - - /* initialize based on the demuxer-supplied streamdata header */ - if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) { - init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8); - metadata_streaminfo(s); - dump_headers(s); - } - - return 0; -} - -static void dump_headers(FLACContext *s) -{ - av_log(s->avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d (%d)\n", s->min_blocksize, s->max_blocksize, s->blocksize); - av_log(s->avctx, AV_LOG_DEBUG, " Framesize: %d .. %d\n", s->min_framesize, s->max_framesize); - av_log(s->avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate); - av_log(s->avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels); - av_log(s->avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps); -} - -static void allocate_buffers(FLACContext *s){ - int i; - - assert(s->max_blocksize); - - if(s->max_framesize == 0 && s->max_blocksize){ - s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; //FIXME header overhead - } - - for (i = 0; i < s->channels; i++) - { - s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t)*s->max_blocksize); - } - - s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize); -} - -static void metadata_streaminfo(FLACContext *s) -{ - /* mandatory streaminfo */ - s->min_blocksize = get_bits(&s->gb, 16); - s->max_blocksize = get_bits(&s->gb, 16); - - s->min_framesize = get_bits_long(&s->gb, 24); - s->max_framesize = get_bits_long(&s->gb, 24); - - s->samplerate = get_bits_long(&s->gb, 20); - s->channels = get_bits(&s->gb, 3) + 1; - s->bps = get_bits(&s->gb, 5) + 1; - - s->avctx->channels = s->channels; - s->avctx->sample_rate = s->samplerate; - - skip_bits(&s->gb, 36); /* total num of samples */ - - skip_bits(&s->gb, 64); /* md5 sum */ - skip_bits(&s->gb, 64); /* md5 sum */ - - allocate_buffers(s); -} - -static int decode_residuals(FLACContext *s, int channel, int pred_order) -{ - int i, tmp, partition, method_type, rice_order; - int sample = 0, samples; - - method_type = get_bits(&s->gb, 2); - if (method_type != 0){ - av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type); + /* frame sync code */ + if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) { + av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n"); return -1; } - rice_order = get_bits(&s->gb, 4); - - samples= s->blocksize >> rice_order; + /* variable block size stream code */ + fi->is_var_size = get_bits1(gb); - sample= - i= pred_order; - for (partition = 0; partition < (1 << rice_order); partition++) - { - tmp = get_bits(&s->gb, 4); - if (tmp == 15) - { - av_log(s->avctx, AV_LOG_DEBUG, "fixed len partition\n"); - tmp = get_bits(&s->gb, 5); - for (; i < samples; i++, sample++) - s->decoded[channel][sample] = get_sbits(&s->gb, tmp); - } - else - { -// av_log(s->avctx, AV_LOG_DEBUG, "rice coded partition k=%d\n", tmp); - for (; i < samples; i++, sample++){ - s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0); - } - } - i= 0; - } + /* block size and sample rate codes */ + bs_code = get_bits(gb, 4); + sr_code = get_bits(gb, 4); -// av_log(s->avctx, AV_LOG_DEBUG, "partitions: %d, samples: %d\n", 1 << rice_order, sample); - - return 0; -} - -static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order) -{ - int i; - -// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME FIXED\n"); - - /* warm up samples */ -// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order); - - for (i = 0; i < pred_order; i++) - { - s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps); -// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]); - } - - if (decode_residuals(s, channel, pred_order) < 0) + /* channels and decorrelation */ + fi->ch_mode = get_bits(gb, 4); + if (fi->ch_mode < FLAC_MAX_CHANNELS) { + fi->channels = fi->ch_mode + 1; + fi->ch_mode = FLAC_CHMODE_INDEPENDENT; + } else if (fi->ch_mode < FLAC_MAX_CHANNELS + FLAC_CHMODE_MID_SIDE) { + fi->channels = 2; + fi->ch_mode -= FLAC_MAX_CHANNELS - 1; + } else { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "invalid channel mode: %d\n", fi->ch_mode); return -1; - - switch(pred_order) - { - case 0: - break; - case 1: - for (i = pred_order; i < s->blocksize; i++) - s->decoded[channel][i] += s->decoded[channel][i-1]; - break; - case 2: - for (i = pred_order; i < s->blocksize; i++) - s->decoded[channel][i] += 2*s->decoded[channel][i-1] - - s->decoded[channel][i-2]; - break; - case 3: - for (i = pred_order; i < s->blocksize; i++) - s->decoded[channel][i] += 3*s->decoded[channel][i-1] - - 3*s->decoded[channel][i-2] - + s->decoded[channel][i-3]; - break; - case 4: - for (i = pred_order; i < s->blocksize; i++) - s->decoded[channel][i] += 4*s->decoded[channel][i-1] - - 6*s->decoded[channel][i-2] - + 4*s->decoded[channel][i-3] - - s->decoded[channel][i-4]; - break; - default: - av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order); - return -1; - } - - return 0; -} - -static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) -{ - int i, j; - int coeff_prec, qlevel; - int coeffs[pred_order]; - -// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME LPC\n"); - - /* warm up samples */ -// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order); - - for (i = 0; i < pred_order; i++) - { - s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps); -// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]); } - coeff_prec = get_bits(&s->gb, 4) + 1; - if (coeff_prec == 16) - { - av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n"); + /* bits per sample */ + bps_code = get_bits(gb, 3); + if (bps_code == 3 || bps_code == 7) { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "invalid sample size code (%d)\n", + bps_code); return -1; } -// av_log(s->avctx, AV_LOG_DEBUG, " qlp coeff prec: %d\n", coeff_prec); - qlevel = get_sbits(&s->gb, 5); -// av_log(s->avctx, AV_LOG_DEBUG, " quant level: %d\n", qlevel); - if(qlevel < 0){ - av_log(s->avctx, AV_LOG_DEBUG, "qlevel %d not supported, maybe buggy stream\n", qlevel); + fi->bps = sample_size_table[bps_code]; + + /* reserved bit */ + if (get_bits1(gb)) { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "broken stream, invalid padding\n"); return -1; } - for (i = 0; i < pred_order; i++) - { - coeffs[i] = get_sbits(&s->gb, coeff_prec); -// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, coeffs[i]); + /* sample or frame count */ + fi->frame_or_sample_num = get_utf8(gb); + if (fi->frame_or_sample_num < 0) { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "sample/frame number invalid; utf8 fscked\n"); + return -1; } - if (decode_residuals(s, channel, pred_order) < 0) + /* blocksize */ + if (bs_code == 0) { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "reserved blocksize code: 0\n"); return -1; - - if (s->bps > 16) { - int64_t sum; - for (i = pred_order; i < s->blocksize; i++) - { - sum = 0; - for (j = 0; j < pred_order; j++) - sum += (int64_t)coeffs[j] * s->decoded[channel][i-j-1]; - s->decoded[channel][i] += sum >> qlevel; - } + } else if (bs_code == 6) { + fi->blocksize = get_bits(gb, 8) + 1; + } else if (bs_code == 7) { + fi->blocksize = get_bits(gb, 16) + 1; } else { - int sum; - for (i = pred_order; i < s->blocksize; i++) - { - sum = 0; - for (j = 0; j < pred_order; j++) - sum += coeffs[j] * s->decoded[channel][i-j-1]; - s->decoded[channel][i] += sum >> qlevel; - } - } - - return 0; -} - -static inline int decode_subframe(FLACContext *s, int channel) -{ - int type, wasted = 0; - int i, tmp; - - s->curr_bps = s->bps; - if(channel == 0){ - if(s->decorrelation == RIGHT_SIDE) - s->curr_bps++; - }else{ - if(s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE) - s->curr_bps++; - } - - if (get_bits1(&s->gb)) - { - av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n"); + fi->blocksize = ff_flac_blocksize_table[bs_code]; + } + + /* sample rate */ + if (sr_code < 12) { + fi->samplerate = ff_flac_sample_rate_table[sr_code]; + } else if (sr_code == 12) { + fi->samplerate = get_bits(gb, 8) * 1000; + } else if (sr_code == 13) { + fi->samplerate = get_bits(gb, 16); + } else if (sr_code == 14) { + fi->samplerate = get_bits(gb, 16) * 10; + } else { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "illegal sample rate code %d\n", + sr_code); return -1; } - type = get_bits(&s->gb, 6); -// wasted = get_bits1(&s->gb); -// if (wasted) -// { -// while (!get_bits1(&s->gb)) -// wasted++; -// if (wasted) -// wasted++; -// s->curr_bps -= wasted; -// } -#if 0 - wasted= 16 - av_log2(show_bits(&s->gb, 17)); - skip_bits(&s->gb, wasted+1); - s->curr_bps -= wasted; -#else - if (get_bits1(&s->gb)) - { - wasted = 1; - while (!get_bits1(&s->gb)) - wasted++; - s->curr_bps -= wasted; - av_log(s->avctx, AV_LOG_DEBUG, "%d wasted bits\n", wasted); - } -#endif -//FIXME use av_log2 for types - if (type == 0) - { - av_log(s->avctx, AV_LOG_DEBUG, "coding type: constant\n"); - tmp = get_sbits(&s->gb, s->curr_bps); - for (i = 0; i < s->blocksize; i++) - s->decoded[channel][i] = tmp; - } - else if (type == 1) - { - av_log(s->avctx, AV_LOG_DEBUG, "coding type: verbatim\n"); - for (i = 0; i < s->blocksize; i++) - s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps); - } - else if ((type >= 8) && (type <= 12)) - { -// av_log(s->avctx, AV_LOG_DEBUG, "coding type: fixed\n"); - if (decode_subframe_fixed(s, channel, type & ~0x8) < 0) - return -1; - } - else if (type >= 32) - { -// av_log(s->avctx, AV_LOG_DEBUG, "coding type: lpc\n"); - if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0) - return -1; - } - else - { - av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n"); + /* header CRC-8 check */ + skip_bits(gb, 8); + if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer, + get_bits_count(gb)/8)) { + av_log(avctx, AV_LOG_ERROR + log_level_offset, + "header crc mismatch\n"); return -1; } - if (wasted) - { - int i; - for (i = 0; i < s->blocksize; i++) - s->decoded[channel][i] <<= wasted; - } - return 0; } -static int decode_frame(FLACContext *s) +int ff_flac_get_max_frame_size(int blocksize, int ch, int bps) { - int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8; - int decorrelation, bps, blocksize, samplerate; - - blocksize_code = get_bits(&s->gb, 4); - - sample_rate_code = get_bits(&s->gb, 4); - - assignment = get_bits(&s->gb, 4); /* channel assignment */ - if (assignment < 8 && s->channels == assignment+1) - decorrelation = INDEPENDENT; - else if (assignment >=8 && assignment < 11 && s->channels == 2) - decorrelation = LEFT_SIDE + assignment - 8; - else - { - av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels); - return -1; - } - - sample_size_code = get_bits(&s->gb, 3); - if(sample_size_code == 0) - bps= s->bps; - else if((sample_size_code != 3) && (sample_size_code != 7)) - bps = sample_size_table[sample_size_code]; - else - { - av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", sample_size_code); - return -1; - } - - if (get_bits1(&s->gb)) - { - av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n"); - return -1; - } - - if(get_utf8(&s->gb) < 0){ - av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n"); - return -1; - } -#if 0 - 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= av_crc(av_crc07, 0, s->gb.buffer, get_bits_count(&s->gb)/8); - if(crc8){ - av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8); - return -1; - } + /* Technically, there is no limit to FLAC frame size, but an encoder + should not write a frame that is larger than if verbatim encoding mode + were to be used. */ - s->blocksize = blocksize; - s->samplerate = samplerate; - s->bps = bps; - s->decorrelation= decorrelation; + int count; -// 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; + count = 16; /* frame header */ + count += ch * ((7+bps+7)/8); /* subframe headers */ + if (ch == 2) { + /* for stereo, need to account for using decorrelation */ + count += (( 2*bps+1) * blocksize + 7) / 8; + } else { + count += ( ch*bps * blocksize + 7) / 8; } + count += 2; /* frame footer */ - align_get_bits(&s->gb); - - /* frame footer */ - skip_bits(&s->gb, 16); /* data crc */ - - return 0; + return count; } -static inline int16_t shift_to_16_bits(int32_t data, int bps) +int avpriv_flac_is_extradata_valid(AVCodecContext *avctx, + enum FLACExtradataFormat *format, + uint8_t **streaminfo_start) { - if (bps == 24) { - return (data >> 8); - } else if (bps == 20) { - return (data >> 4); + if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) { + av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n"); + return 0; + } + if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) { + /* extradata contains STREAMINFO only */ + if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) { + av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n", + FLAC_STREAMINFO_SIZE-avctx->extradata_size); + } + *format = FLAC_EXTRADATA_FORMAT_STREAMINFO; + *streaminfo_start = avctx->extradata; } else { - return data; + if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) { + av_log(avctx, AV_LOG_ERROR, "extradata too small.\n"); + return 0; + } + *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER; + *streaminfo_start = &avctx->extradata[8]; } + return 1; } -static int flac_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - uint8_t *buf, int buf_size) +void ff_flac_set_channel_layout(AVCodecContext *avctx) { - 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= 65536; // 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= FFMAX(FFMIN(buf_size, s->max_framesize - s->bitstream_size), 0); - 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); - - /* Buffer might have been reallocated, reinit bitreader */ - if(buf != &s->bitstream[s->bitstream_index]) - { - int bits_count = get_bits_count(&s->gb); - buf= &s->bitstream[s->bitstream_index]; - init_get_bits(&s->gb, buf, buf_size*8); - skip_bits(&s->gb, bits_count); - } - - dump_headers(s); - break;} - default: - for(i=0; igb, 8); - } - } - } while(!metadata_last); - } + if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts)) + avctx->channel_layout = flac_channel_layouts[avctx->channels - 1]; 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; - } - } - + avctx->channel_layout = 0; +} -#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]; +void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, + const uint8_t *buffer) +{ + GetBitContext gb; + init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8); - right[i] = x - (y / 2); - left[i] = right[i] + y; - } - *data_size = 2 * s->blocksize; + skip_bits(&gb, 16); /* skip min blocksize */ + s->max_blocksize = get_bits(&gb, 16); + if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) { + av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n", + s->max_blocksize); + s->max_blocksize = 16; } - 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++) = shift_to_16_bits(s->decoded[i][j], s->bps); - } - break; - case LEFT_SIDE: - assert(s->channels == 2); - for (i = 0; i < s->blocksize; i++) - { - *(samples++) = shift_to_16_bits(s->decoded[0][i], s->bps); - *(samples++) = shift_to_16_bits(s->decoded[0][i] - - s->decoded[1][i], s->bps); - } - break; - case RIGHT_SIDE: - assert(s->channels == 2); - for (i = 0; i < s->blocksize; i++) - { - *(samples++) = shift_to_16_bits(s->decoded[0][i] - + s->decoded[1][i], s->bps); - *(samples++) = shift_to_16_bits(s->decoded[1][i], s->bps); - } - 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++) = shift_to_16_bits(mid + side, s->bps); - *(samples++) = shift_to_16_bits(mid, s->bps); -#else + skip_bits(&gb, 24); /* skip min frame size */ + s->max_framesize = get_bits_long(&gb, 24); - mid <<= 1; - if (side & 1) - mid++; - *(samples++) = (mid + side) >> 1; - *(samples++) = (mid - side) >> 1; -#endif - } - break; - } -#endif + s->samplerate = get_bits_long(&gb, 20); + s->channels = get_bits(&gb, 3) + 1; + s->bps = get_bits(&gb, 5) + 1; - *data_size = (int8_t *)samples - (int8_t *)data; -// av_log(s->avctx, AV_LOG_DEBUG, "data size: %d\n", *data_size); + avctx->channels = s->channels; + avctx->sample_rate = s->samplerate; + avctx->bits_per_raw_sample = s->bps; + ff_flac_set_channel_layout(avctx); -// 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; - } + s->samples = get_bits_long(&gb, 32) << 4; + s->samples |= get_bits(&gb, 4); - if(s->bitstream_size){ - s->bitstream_index += i; - s->bitstream_size -= i; - return input_buf_size; - }else - return i; + skip_bits_long(&gb, 64); /* md5 sum */ + skip_bits_long(&gb, 64); /* md5 sum */ } -static int flac_decode_close(AVCodecContext *avctx) +void avpriv_flac_parse_block_header(const uint8_t *block_header, + int *last, int *type, int *size) { - 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; + int tmp = bytestream_get_byte(&block_header); + if (last) + *last = tmp & 0x80; + if (type) + *type = tmp & 0x7F; + if (size) + *size = bytestream_get_be24(&block_header); } - -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, -};