diff options
Diffstat (limited to 'contrib/ffmpeg/libavcodec/flac.c')
| -rw-r--r-- | contrib/ffmpeg/libavcodec/flac.c | 752 |
1 files changed, 752 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavcodec/flac.c b/contrib/ffmpeg/libavcodec/flac.c new file mode 100644 index 000000000..6c64ad0a1 --- /dev/null +++ b/contrib/ffmpeg/libavcodec/flac.c @@ -0,0 +1,752 @@ +/* + * 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 "avcodec.h" +#include "bitstream.h" +#include "golomb.h" +#include "crc.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 { + 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_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); + streaminfo_updated = 1; + break; + + 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 != 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; + + 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) +{ + 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) + 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"); + 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 { + 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"); + 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 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; + } + + 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 inline int16_t shift_to_16_bits(int32_t data, int bps) +{ + if (bps == 24) { + return (data >> 8); + } else if (bps == 20) { + return (data >> 4); + } else { + return data; + } +} + +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; + + 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) < 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 ) >> (16 - s->bps);\ + *(samples++) = (right) >> (16 - s->bps);\ + }\ + break; + + 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: + 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, +}; |