/* * 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 libavcodec/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 #include "libavutil/crc.h" #include "avcodec.h" #include "internal.h" #include "bitstream.h" #include "bytestream.h" #include "golomb.h" #include "flac.h" #undef NDEBUG #include #define MAX_CHANNELS 8 #define MAX_BLOCKSIZE 65535 enum decorrelation_type { INDEPENDENT, LEFT_SIDE, RIGHT_SIDE, MID_SIDE, }; typedef struct FLACContext { FLACSTREAMINFO AVCodecContext *avctx; ///< parent AVCodecContext GetBitContext gb; ///< GetBitContext initialized to start at the current frame int blocksize; ///< number of samples in the current frame int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits int sample_shift; ///< shift required to make output samples 16-bit or 32-bit int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit enum decorrelation_type decorrelation; ///< channel decorrelation type in the current frame int got_streaminfo; ///< indicates if the STREAMINFO has been read int32_t *decoded[MAX_CHANNELS]; ///< decoded samples uint8_t *bitstream; unsigned int bitstream_size; unsigned int bitstream_index; unsigned int allocated_bitstream_size; } FLACContext; 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); int ff_flac_is_extradata_valid(AVCodecContext *avctx, enum FLACExtradataFormat *format, uint8_t **streaminfo_start) { 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 { 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 av_cold int flac_decode_init(AVCodecContext *avctx) { enum FLACExtradataFormat format; uint8_t *streaminfo; FLACContext *s = avctx->priv_data; s->avctx = avctx; avctx->sample_fmt = SAMPLE_FMT_S16; /* for now, the raw FLAC header is allowed to be passed to the decoder as frame data instead of extradata. */ if (!avctx->extradata) return 0; if (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo)) return -1; /* initialize based on the demuxer-supplied streamdata header */ ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo); allocate_buffers(s); s->got_streaminfo = 1; return 0; } static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) { av_log(avctx, AV_LOG_DEBUG, " Max Blocksize: %d\n", 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) { // FIXME header overhead s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; } 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); skip_bits(&gb, 16); /* skip min blocksize */ s->max_blocksize = get_bits(&gb, 16); if (s->max_blocksize < 16) { av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n", s->max_blocksize); s->max_blocksize = 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; avctx->bits_per_raw_sample = s->bps; if (s->bps > 16) avctx->sample_fmt = SAMPLE_FMT_S32; else avctx->sample_fmt = SAMPLE_FMT_S16; s->samples = get_bits_long(&gb, 32) << 4; s->samples |= get_bits(&gb, 4); skip_bits_long(&gb, 64); /* md5 sum */ skip_bits_long(&gb, 64); /* md5 sum */ dump_headers(avctx, s); } /** * Parse the STREAMINFO from an inline header. * @param s the flac decoding context * @param buf input buffer, starting with the "fLaC" marker * @param buf_size buffer size * @return non-zero if metadata is invalid */ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) { int metadata_type, metadata_size; if (buf_size < FLAC_STREAMINFO_SIZE+8) { /* need more data */ return 0; } buf += 4; metadata_type = bytestream_get_byte(&buf) & 0x7F; metadata_size = bytestream_get_be24(&buf); if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO || metadata_size != FLAC_STREAMINFO_SIZE) { return AVERROR_INVALIDDATA; } ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, buf); allocate_buffers(s); s->got_streaminfo = 1; return 0; } /** * Determine the size of an inline header. * @param buf input buffer, starting with the "fLaC" marker * @param buf_size buffer size * @return number of bytes in the header, or 0 if more data is needed */ static int get_metadata_size(const uint8_t *buf, int buf_size) { int metadata_last, metadata_size; const uint8_t *buf_end = buf + buf_size; buf += 4; do { metadata_last = bytestream_get_byte(&buf) & 0x80; metadata_size = bytestream_get_be24(&buf); if (buf + metadata_size > buf_end) { /* need more data in order to read the complete header */ return 0; } buf += metadata_size; } while (!metadata_last); return buf_size - (buf_end - buf); } 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_ERROR, "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)) { tmp = get_bits(&s->gb, 5); for (; i < samples; i++, sample++) s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp); } else { for (; i < samples; i++, sample++) { s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0); } } i= 0; } 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 av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i; /* warm up samples */ for (i = 0; i < pred_order; i++) { decoded[i] = get_sbits_long(&s->gb, s->curr_bps); } 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]; /* warm up samples */ for (i = 0; i < pred_order; i++) { decoded[i] = get_sbits_long(&s->gb, s->curr_bps); } coeff_prec = get_bits(&s->gb, 4) + 1; if (coeff_prec == 16) { av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n"); return -1; } qlevel = get_sbits(&s->gb, 5); if (qlevel < 0) { av_log(s->avctx, AV_LOG_ERROR, "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); } 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); if (get_bits1(&s->gb)) { wasted = 1; while (!get_bits1(&s->gb)) wasted++; s->curr_bps -= wasted; } if (s->curr_bps > 32) { ff_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0); return -1; } //FIXME use av_log2 for types if (type == 0) { tmp = get_sbits_long(&s->gb, s->curr_bps); for (i = 0; i < s->blocksize; i++) s->decoded[channel][i] = tmp; } else if (type == 1) { for (i = 0; i < s->blocksize; i++) s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps); } else if ((type >= 8) && (type <= 12)) { if (decode_subframe_fixed(s, channel, type & ~0x8) < 0) return -1; } else if (type >= 32) { 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 (bps > 16) { s->avctx->sample_fmt = SAMPLE_FMT_S32; s->sample_shift = 32 - bps; s->is32 = 1; } else { s->avctx->sample_fmt = SAMPLE_FMT_S16; s->sample_shift = 16 - bps; s->is32 = 0; } s->bps = s->avctx->bits_per_raw_sample = bps; 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 (blocksize_code == 0) { av_log(s->avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n"); return -1; } 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 * (s->is32 ? 4 : 2) > 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++) { 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 i, j = 0, input_buf_size = 0, bytes_read = 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; } } /* check that there is at least the smallest decodable amount of data. this amount corresponds to the smallest valid FLAC frame possible. */ if (buf_size < 11) goto end; /* check for inline header */ if (AV_RB32(buf) == MKBETAG('f','L','a','C')) { if (!s->got_streaminfo && parse_streaminfo(s, buf, buf_size)) { av_log(s->avctx, AV_LOG_ERROR, "invalid header\n"); return -1; } bytes_read = get_metadata_size(buf, buf_size); goto end; } /* check for frame sync code and resync stream if necessary */ if ((AV_RB16(buf) & 0xFFFE) != 0xFFF8) { const uint8_t *buf_end = buf + buf_size; av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n"); while (buf+2 < buf_end && (AV_RB16(buf) & 0xFFFE) != 0xFFF8) buf++; bytes_read = buf_size - (buf_end - buf); goto end; // we may not have enough bits left to decode a frame, so try next time } /* decode frame */ init_get_bits(&s->gb, buf, buf_size*8); 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; } *data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2); bytes_read = (get_bits_count(&s->gb)+7)/8; #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) } end: if (bytes_read > buf_size) { av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size); s->bitstream_size=0; s->bitstream_index=0; return -1; } if (s->bitstream_size) { s->bitstream_index += bytes_read; s->bitstream_size -= bytes_read; return input_buf_size; } else return bytes_read; } 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)"), };