/* * 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 #define ALT_BITSTREAM_READER #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 }; static int64_t get_utf8(GetBitContext *gb){ int64_t val; GET_UTF8(val, get_bits(gb, 8), return -1;) return val; } static void metadata_streaminfo(FLACContext *s); static void allocate_buffers(FLACContext *s); static int metadata_parse(FLACContext *s); static 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 */ init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8); if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) { metadata_streaminfo(s); allocate_buffers(s); } else { metadata_parse(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 */ dump_headers(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; 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); 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); streaminfo_updated = 1; break; default: for (i=0; igb, 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 != 0){ 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, 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; } // 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; a = decoded[pred_order-1]; b = a - decoded[pred_order-2]; c = b - decoded[pred_order-2] + decoded[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]; // 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"); 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] * s->decoded[channel][i-j-1]; s->decoded[channel][i] += sum >> qlevel; } } else { for (i = pred_order; i < s->blocksize-1; i += 2) { int c = coeffs[pred_order-1]; int s0 = c * s->decoded[channel][i-pred_order]; int s1 = 0; for (j = pred_order-1; j > 0; j--) { int d = s->decoded[channel][i-j]; s1 += c*d; c = coeffs[j-1]; s0 += c*d; } s0 = s->decoded[channel][i] += s0 >> qlevel; s1 += c * s0; s->decoded[channel][i+1] += s1 >> qlevel; } if (i < s->blocksize) { int 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"); 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 > 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; } s->blocksize = blocksize; s->samplerate = samplerate; s->bps = bps; s->decorrelation= decorrelation; // dump_headers(s); /* subframes */ for (i = 0; i < s->channels; i++) { // av_log(s->avctx, AV_LOG_DEBUG, "decoded: %x residual: %x\n", s->decoded[i], s->residual[i]); if (decode_subframe(s, i) < 0) return -1; } align_get_bits(&s->gb); /* frame footer */ skip_bits(&s->gb, 16); /* data crc */ return 0; } static int flac_decode_frame(AVCodecContext *avctx, void *data, int *data_size, uint8_t *buf, int buf_size) { FLACContext *s = avctx->priv_data; int 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= 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); if (!metadata_parse(s)) { 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, 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 int flac_decode_close(AVCodecContext *avctx) { FLACContext *s = avctx->priv_data; int i; for (i = 0; i < s->channels; i++) { av_freep(&s->decoded[i]); } av_freep(&s->bitstream); return 0; } static void flac_flush(AVCodecContext *avctx){ FLACContext *s = avctx->priv_data; s->bitstream_size= s->bitstream_index= 0; } AVCodec flac_decoder = { "flac", CODEC_TYPE_AUDIO, CODEC_ID_FLAC, sizeof(FLACContext), flac_decode_init, NULL, flac_decode_close, flac_decode_frame, .flush= flac_flush, };