Mercurial > libavcodec.hg
changeset 8642:178d5dfccad4 libavcodec
rename flac.c to flacdec.c
| author | jbr |
|---|---|
| date | Fri, 23 Jan 2009 22:27:19 +0000 |
| parents | 54e2916c25a5 |
| children | 1fa3e8a72ca5 |
| files | Makefile flac.c flacdec.c |
| diffstat | 3 files changed, 796 insertions(+), 796 deletions(-) [+] |
line wrap: on
line diff
--- a/Makefile Fri Jan 23 12:09:32 2009 +0000 +++ b/Makefile Fri Jan 23 22:27:19 2009 +0000 @@ -82,7 +82,7 @@ OBJS-$(CONFIG_FFV1_ENCODER) += ffv1.o rangecoder.o OBJS-$(CONFIG_FFVHUFF_DECODER) += huffyuv.o OBJS-$(CONFIG_FFVHUFF_ENCODER) += huffyuv.o -OBJS-$(CONFIG_FLAC_DECODER) += flac.o +OBJS-$(CONFIG_FLAC_DECODER) += flacdec.o OBJS-$(CONFIG_FLAC_ENCODER) += flacenc.o lpc.o OBJS-$(CONFIG_FLASHSV_DECODER) += flashsv.o OBJS-$(CONFIG_FLASHSV_ENCODER) += flashsvenc.o
--- a/flac.c Fri Jan 23 12:09:32 2009 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,795 +0,0 @@ -/* - * FLAC (Free Lossless Audio Codec) decoder - * Copyright (c) 2003 Alex Beregszaszi - * - * This file is part of FFmpeg. - * - * FFmpeg 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 2.1 of the License, or (at your option) any later version. - * - * FFmpeg 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 FFmpeg; 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 <limits.h> - -#define ALT_BITSTREAM_READER -#include "libavutil/crc.h" -#include "avcodec.h" -#include "bitstream.h" -#include "golomb.h" -#include "flac.h" - -#undef NDEBUG -#include <assert.h> - -#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 { - FLACSTREAMINFO - - AVCodecContext *avctx; - GetBitContext gb; - - int blocksize/*, last_blocksize*/; - int curr_bps; - enum decorrelation_type decorrelation; - - int32_t *decoded[MAX_CHANNELS]; - uint8_t *bitstream; - unsigned int bitstream_size; - unsigned int bitstream_index; - unsigned int allocated_bitstream_size; -} FLACContext; - -#define METADATA_TYPE_STREAMINFO 0 - -static const int sample_rate_table[] = -{ 0, - 88200, 176400, 192000, - 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000, - 0, 0, 0, 0 }; - -static const int sample_size_table[] = -{ 0, 8, 12, 0, 16, 20, 24, 0 }; - -static const 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 -}; - -static int64_t get_utf8(GetBitContext *gb){ - int64_t val; - GET_UTF8(val, get_bits(gb, 8), return -1;) - return val; -} - -static void allocate_buffers(FLACContext *s); -static int metadata_parse(FLACContext *s); - -static av_cold int flac_decode_init(AVCodecContext * avctx) -{ - FLACContext *s = avctx->priv_data; - s->avctx = avctx; - - if (avctx->extradata_size > 4) { - /* initialize based on the demuxer-supplied streamdata header */ - if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) { - ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, avctx->extradata); - allocate_buffers(s); - } else { - init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8); - metadata_parse(s); - } - } - - avctx->sample_fmt = SAMPLE_FMT_S16; - return 0; -} - -static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) -{ - av_log(avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d\n", s->min_blocksize, s->max_blocksize); - av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize); - av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate); - av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels); - av_log(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); - } - - if(s->allocated_bitstream_size < s->max_framesize) - s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize); -} - -void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, - const uint8_t *buffer) -{ - GetBitContext gb; - init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8); - - /* mandatory streaminfo */ - s->min_blocksize = get_bits(&gb, 16); - s->max_blocksize = get_bits(&gb, 16); - - skip_bits(&gb, 24); /* skip min frame size */ - s->max_framesize = get_bits_long(&gb, 24); - - s->samplerate = get_bits_long(&gb, 20); - s->channels = get_bits(&gb, 3) + 1; - s->bps = get_bits(&gb, 5) + 1; - - avctx->channels = s->channels; - avctx->sample_rate = s->samplerate; - - skip_bits(&gb, 36); /* total num of samples */ - - skip_bits(&gb, 64); /* md5 sum */ - skip_bits(&gb, 64); /* md5 sum */ - - dump_headers(avctx, s); -} - -/** - * Parse a list of metadata blocks. This list of blocks must begin with - * the fLaC marker. - * @param s the flac decoding context containing the gb bit reader used to - * parse metadata - * @return 1 if some metadata was read, 0 if no fLaC marker was found - */ -static int metadata_parse(FLACContext *s) -{ - int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0; - int initial_pos= get_bits_count(&s->gb); - - if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) { - skip_bits(&s->gb, 32); - - av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n"); - do { - metadata_last = get_bits1(&s->gb); - metadata_type = get_bits(&s->gb, 7); - metadata_size = get_bits_long(&s->gb, 24); - - if(get_bits_count(&s->gb) + 8*metadata_size > s->gb.size_in_bits){ - skip_bits_long(&s->gb, initial_pos - get_bits_count(&s->gb)); - break; - } - - 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: - ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, s->gb.buffer+get_bits_count(&s->gb)/8); - streaminfo_updated = 1; - - default: - for (i=0; i<metadata_size; i++) - skip_bits(&s->gb, 8); - } - } - } while (!metadata_last); - - if (streaminfo_updated) - allocate_buffers(s); - return 1; - } - return 0; -} - -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 > 1){ - av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type); - return -1; - } - - rice_order = get_bits(&s->gb, 4); - - samples= s->blocksize >> rice_order; - if (pred_order > samples) { - av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", pred_order, samples); - return -1; - } - - sample= - i= pred_order; - for (partition = 0; partition < (1 << rice_order); partition++) - { - tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5); - if (tmp == (method_type == 0 ? 15 : 31)) - { - 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; - } - -// 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) -{ - const int blocksize = s->blocksize; - int32_t *decoded = s->decoded[channel]; - int a, b, c, d, 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++) - { - decoded[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) - return -1; - - if(pred_order > 0) - a = decoded[pred_order-1]; - if(pred_order > 1) - b = a - decoded[pred_order-2]; - if(pred_order > 2) - c = b - decoded[pred_order-2] + decoded[pred_order-3]; - if(pred_order > 3) - d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4]; - - switch(pred_order) - { - case 0: - break; - case 1: - for (i = pred_order; i < blocksize; i++) - decoded[i] = a += decoded[i]; - break; - case 2: - for (i = pred_order; i < blocksize; i++) - decoded[i] = a += b += decoded[i]; - break; - case 3: - for (i = pred_order; i < blocksize; i++) - decoded[i] = a += b += c += decoded[i]; - break; - case 4: - for (i = pred_order; i < blocksize; i++) - decoded[i] = a += b += c += d += decoded[i]; - 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]; - int32_t *decoded = s->decoded[channel]; - -// 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++) - { - decoded[i] = get_sbits(&s->gb, s->curr_bps); -// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, decoded[i]); - } - - coeff_prec = get_bits(&s->gb, 4) + 1; - if (coeff_prec == 16) - { - av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n"); - 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); - 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]); - } - - if (decode_residuals(s, channel, pred_order) < 0) - 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] * decoded[i-j-1]; - decoded[i] += sum >> qlevel; - } - } else { - for (i = pred_order; i < s->blocksize-1; i += 2) - { - int c; - int d = decoded[i-pred_order]; - int s0 = 0, s1 = 0; - for (j = pred_order-1; j > 0; j--) - { - c = coeffs[j]; - s0 += c*d; - d = decoded[i-j]; - s1 += c*d; - } - c = coeffs[0]; - s0 += c*d; - d = decoded[i] += s0 >> qlevel; - s1 += c*d; - decoded[i+1] += s1 >> qlevel; - } - if (i < s->blocksize) - { - int sum = 0; - for (j = 0; j < pred_order; j++) - sum += coeffs[j] * decoded[i-j-1]; - decoded[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"); - 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"); - 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 alloc_data_size) -{ - 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(blocksize * s->channels * sizeof(int16_t) > alloc_data_size) - return -1; - - if (sample_rate_code == 0){ - samplerate= s->samplerate; - }else if (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_crc_get_table(AV_CRC_8_ATM), 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; - } - - s->blocksize = blocksize; - s->samplerate = samplerate; - s->bps = bps; - s->decorrelation= decorrelation; - -// dump_headers(s->avctx, (FLACStreaminfo *)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, - 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 = 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){ -// 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 && input_buf_size){ -// printf("wanna more data ...\n"); - 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; - } - - -#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 -#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];\ - *samples++ = ((left) << (24 - s->bps)) >> 8;\ - *samples++ = ((right) << (24 - s->bps)) >> 8;\ - }\ - break; - - switch(s->decorrelation) - { - case INDEPENDENT: - for (j = 0; j < s->blocksize; j++) - { - for (i = 0; i < s->channels; i++) - *samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8; - } - break; - case LEFT_SIDE: - DECORRELATE(a,a-b) - case RIGHT_SIDE: - DECORRELATE(a+b,b) - case MID_SIDE: - DECORRELATE( (a-=b>>1) + b, a) - } -#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 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)"), -};
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/flacdec.c Fri Jan 23 22:27:19 2009 +0000 @@ -0,0 +1,795 @@ +/* + * FLAC (Free Lossless Audio Codec) decoder + * Copyright (c) 2003 Alex Beregszaszi + * + * This file is part of FFmpeg. + * + * FFmpeg 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 2.1 of the License, or (at your option) any later version. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file flacdec.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 <limits.h> + +#define ALT_BITSTREAM_READER +#include "libavutil/crc.h" +#include "avcodec.h" +#include "bitstream.h" +#include "golomb.h" +#include "flac.h" + +#undef NDEBUG +#include <assert.h> + +#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 { + FLACSTREAMINFO + + AVCodecContext *avctx; + GetBitContext gb; + + int blocksize/*, last_blocksize*/; + int curr_bps; + enum decorrelation_type decorrelation; + + int32_t *decoded[MAX_CHANNELS]; + uint8_t *bitstream; + unsigned int bitstream_size; + unsigned int bitstream_index; + unsigned int allocated_bitstream_size; +} FLACContext; + +#define METADATA_TYPE_STREAMINFO 0 + +static const int sample_rate_table[] = +{ 0, + 88200, 176400, 192000, + 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000, + 0, 0, 0, 0 }; + +static const int sample_size_table[] = +{ 0, 8, 12, 0, 16, 20, 24, 0 }; + +static const 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 +}; + +static int64_t get_utf8(GetBitContext *gb){ + int64_t val; + GET_UTF8(val, get_bits(gb, 8), return -1;) + return val; +} + +static void allocate_buffers(FLACContext *s); +static int metadata_parse(FLACContext *s); + +static av_cold int flac_decode_init(AVCodecContext * avctx) +{ + FLACContext *s = avctx->priv_data; + s->avctx = avctx; + + if (avctx->extradata_size > 4) { + /* initialize based on the demuxer-supplied streamdata header */ + if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) { + ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, avctx->extradata); + allocate_buffers(s); + } else { + init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8); + metadata_parse(s); + } + } + + avctx->sample_fmt = SAMPLE_FMT_S16; + return 0; +} + +static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) +{ + av_log(avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d\n", s->min_blocksize, s->max_blocksize); + av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize); + av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate); + av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels); + av_log(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); + } + + if(s->allocated_bitstream_size < s->max_framesize) + s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize); +} + +void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, + const uint8_t *buffer) +{ + GetBitContext gb; + init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8); + + /* mandatory streaminfo */ + s->min_blocksize = get_bits(&gb, 16); + s->max_blocksize = get_bits(&gb, 16); + + skip_bits(&gb, 24); /* skip min frame size */ + s->max_framesize = get_bits_long(&gb, 24); + + s->samplerate = get_bits_long(&gb, 20); + s->channels = get_bits(&gb, 3) + 1; + s->bps = get_bits(&gb, 5) + 1; + + avctx->channels = s->channels; + avctx->sample_rate = s->samplerate; + + skip_bits(&gb, 36); /* total num of samples */ + + skip_bits(&gb, 64); /* md5 sum */ + skip_bits(&gb, 64); /* md5 sum */ + + dump_headers(avctx, s); +} + +/** + * Parse a list of metadata blocks. This list of blocks must begin with + * the fLaC marker. + * @param s the flac decoding context containing the gb bit reader used to + * parse metadata + * @return 1 if some metadata was read, 0 if no fLaC marker was found + */ +static int metadata_parse(FLACContext *s) +{ + int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0; + int initial_pos= get_bits_count(&s->gb); + + if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) { + skip_bits(&s->gb, 32); + + av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n"); + do { + metadata_last = get_bits1(&s->gb); + metadata_type = get_bits(&s->gb, 7); + metadata_size = get_bits_long(&s->gb, 24); + + if(get_bits_count(&s->gb) + 8*metadata_size > s->gb.size_in_bits){ + skip_bits_long(&s->gb, initial_pos - get_bits_count(&s->gb)); + break; + } + + 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: + ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, s->gb.buffer+get_bits_count(&s->gb)/8); + streaminfo_updated = 1; + + default: + for (i=0; i<metadata_size; i++) + skip_bits(&s->gb, 8); + } + } + } while (!metadata_last); + + if (streaminfo_updated) + allocate_buffers(s); + return 1; + } + return 0; +} + +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 > 1){ + av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type); + return -1; + } + + rice_order = get_bits(&s->gb, 4); + + samples= s->blocksize >> rice_order; + if (pred_order > samples) { + av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", pred_order, samples); + return -1; + } + + sample= + i= pred_order; + for (partition = 0; partition < (1 << rice_order); partition++) + { + tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5); + if (tmp == (method_type == 0 ? 15 : 31)) + { + 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; + } + +// 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) +{ + const int blocksize = s->blocksize; + int32_t *decoded = s->decoded[channel]; + int a, b, c, d, 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++) + { + decoded[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) + return -1; + + if(pred_order > 0) + a = decoded[pred_order-1]; + if(pred_order > 1) + b = a - decoded[pred_order-2]; + if(pred_order > 2) + c = b - decoded[pred_order-2] + decoded[pred_order-3]; + if(pred_order > 3) + d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4]; + + switch(pred_order) + { + case 0: + break; + case 1: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += decoded[i]; + break; + case 2: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += b += decoded[i]; + break; + case 3: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += b += c += decoded[i]; + break; + case 4: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += b += c += d += decoded[i]; + 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]; + int32_t *decoded = s->decoded[channel]; + +// 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++) + { + decoded[i] = get_sbits(&s->gb, s->curr_bps); +// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, decoded[i]); + } + + coeff_prec = get_bits(&s->gb, 4) + 1; + if (coeff_prec == 16) + { + av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n"); + 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); + 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]); + } + + if (decode_residuals(s, channel, pred_order) < 0) + 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] * decoded[i-j-1]; + decoded[i] += sum >> qlevel; + } + } else { + for (i = pred_order; i < s->blocksize-1; i += 2) + { + int c; + int d = decoded[i-pred_order]; + int s0 = 0, s1 = 0; + for (j = pred_order-1; j > 0; j--) + { + c = coeffs[j]; + s0 += c*d; + d = decoded[i-j]; + s1 += c*d; + } + c = coeffs[0]; + s0 += c*d; + d = decoded[i] += s0 >> qlevel; + s1 += c*d; + decoded[i+1] += s1 >> qlevel; + } + if (i < s->blocksize) + { + int sum = 0; + for (j = 0; j < pred_order; j++) + sum += coeffs[j] * decoded[i-j-1]; + decoded[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"); + 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"); + 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 alloc_data_size) +{ + 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(blocksize * s->channels * sizeof(int16_t) > alloc_data_size) + return -1; + + if (sample_rate_code == 0){ + samplerate= s->samplerate; + }else if (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_crc_get_table(AV_CRC_8_ATM), 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; + } + + s->blocksize = blocksize; + s->samplerate = samplerate; + s->bps = bps; + s->decorrelation= decorrelation; + +// dump_headers(s->avctx, (FLACStreaminfo *)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, + 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 = 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){ +// 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 && input_buf_size){ +// printf("wanna more data ...\n"); + 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; + } + + +#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 +#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];\ + *samples++ = ((left) << (24 - s->bps)) >> 8;\ + *samples++ = ((right) << (24 - s->bps)) >> 8;\ + }\ + break; + + switch(s->decorrelation) + { + case INDEPENDENT: + for (j = 0; j < s->blocksize; j++) + { + for (i = 0; i < s->channels; i++) + *samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8; + } + break; + case LEFT_SIDE: + DECORRELATE(a,a-b) + case RIGHT_SIDE: + DECORRELATE(a+b,b) + case MID_SIDE: + DECORRELATE( (a-=b>>1) + b, a) + } +#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 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)"), +};