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671 lines
20 KiB
671 lines
20 KiB
/* |
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* FLAC (Free Lossless Audio Codec) decoder |
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* Copyright (c) 2003 Alex Beregszaszi |
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* |
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* This file is part of Libav. |
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* |
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* Libav is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* Libav is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with Libav; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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|
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/** |
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* @file |
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* FLAC (Free Lossless Audio Codec) decoder |
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* @author Alex Beregszaszi |
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* @see http://flac.sourceforge.net/ |
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* |
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* This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed |
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* through, starting from the initial 'fLaC' signature; or by passing the |
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* 34-byte streaminfo structure through avctx->extradata[_size] followed |
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* by data starting with the 0xFFF8 marker. |
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*/ |
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|
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#include <limits.h> |
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|
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#include "libavutil/crc.h" |
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#include "avcodec.h" |
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#include "internal.h" |
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#include "get_bits.h" |
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#include "bytestream.h" |
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#include "golomb.h" |
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#include "flac.h" |
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#include "flacdata.h" |
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|
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#undef NDEBUG |
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#include <assert.h> |
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|
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typedef struct FLACContext { |
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FLACSTREAMINFO |
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AVCodecContext *avctx; ///< parent AVCodecContext |
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AVFrame frame; |
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GetBitContext gb; ///< GetBitContext initialized to start at the current frame |
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|
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int blocksize; ///< number of samples in the current frame |
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int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits |
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int sample_shift; ///< shift required to make output samples 16-bit or 32-bit |
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int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit |
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int ch_mode; ///< channel decorrelation type in the current frame |
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int got_streaminfo; ///< indicates if the STREAMINFO has been read |
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int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples |
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} FLACContext; |
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|
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static void allocate_buffers(FLACContext *s); |
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|
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int avpriv_flac_is_extradata_valid(AVCodecContext *avctx, |
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enum FLACExtradataFormat *format, |
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uint8_t **streaminfo_start) |
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{ |
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if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) { |
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av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n"); |
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return 0; |
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} |
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if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) { |
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/* extradata contains STREAMINFO only */ |
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if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) { |
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av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n", |
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FLAC_STREAMINFO_SIZE-avctx->extradata_size); |
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} |
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*format = FLAC_EXTRADATA_FORMAT_STREAMINFO; |
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*streaminfo_start = avctx->extradata; |
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} else { |
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if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) { |
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av_log(avctx, AV_LOG_ERROR, "extradata too small.\n"); |
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return 0; |
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} |
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*format = FLAC_EXTRADATA_FORMAT_FULL_HEADER; |
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*streaminfo_start = &avctx->extradata[8]; |
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} |
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return 1; |
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} |
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|
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static av_cold int flac_decode_init(AVCodecContext *avctx) |
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{ |
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enum FLACExtradataFormat format; |
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uint8_t *streaminfo; |
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FLACContext *s = avctx->priv_data; |
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s->avctx = avctx; |
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avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
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|
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/* for now, the raw FLAC header is allowed to be passed to the decoder as |
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frame data instead of extradata. */ |
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if (!avctx->extradata) |
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return 0; |
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if (!avpriv_flac_is_extradata_valid(avctx, &format, &streaminfo)) |
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return -1; |
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|
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/* initialize based on the demuxer-supplied streamdata header */ |
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avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo); |
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if (s->bps > 16) |
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avctx->sample_fmt = AV_SAMPLE_FMT_S32; |
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else |
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avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
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allocate_buffers(s); |
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s->got_streaminfo = 1; |
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avcodec_get_frame_defaults(&s->frame); |
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avctx->coded_frame = &s->frame; |
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return 0; |
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} |
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static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) |
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{ |
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av_log(avctx, AV_LOG_DEBUG, " Max Blocksize: %d\n", s->max_blocksize); |
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av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize); |
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av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate); |
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av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels); |
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av_log(avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps); |
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} |
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static void allocate_buffers(FLACContext *s) |
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{ |
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int i; |
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assert(s->max_blocksize); |
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for (i = 0; i < s->channels; i++) { |
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s->decoded[i] = av_realloc(s->decoded[i], |
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sizeof(int32_t)*s->max_blocksize); |
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} |
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} |
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void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, |
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const uint8_t *buffer) |
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{ |
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GetBitContext gb; |
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init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8); |
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skip_bits(&gb, 16); /* skip min blocksize */ |
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s->max_blocksize = get_bits(&gb, 16); |
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if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) { |
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av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n", |
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s->max_blocksize); |
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s->max_blocksize = 16; |
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} |
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skip_bits(&gb, 24); /* skip min frame size */ |
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s->max_framesize = get_bits_long(&gb, 24); |
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s->samplerate = get_bits_long(&gb, 20); |
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s->channels = get_bits(&gb, 3) + 1; |
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s->bps = get_bits(&gb, 5) + 1; |
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avctx->channels = s->channels; |
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avctx->sample_rate = s->samplerate; |
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avctx->bits_per_raw_sample = s->bps; |
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s->samples = get_bits_long(&gb, 32) << 4; |
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s->samples |= get_bits(&gb, 4); |
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skip_bits_long(&gb, 64); /* md5 sum */ |
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skip_bits_long(&gb, 64); /* md5 sum */ |
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dump_headers(avctx, s); |
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} |
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void avpriv_flac_parse_block_header(const uint8_t *block_header, |
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int *last, int *type, int *size) |
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{ |
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int tmp = bytestream_get_byte(&block_header); |
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if (last) |
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*last = tmp & 0x80; |
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if (type) |
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*type = tmp & 0x7F; |
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if (size) |
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*size = bytestream_get_be24(&block_header); |
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} |
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/** |
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* Parse the STREAMINFO from an inline header. |
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* @param s the flac decoding context |
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* @param buf input buffer, starting with the "fLaC" marker |
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* @param buf_size buffer size |
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* @return non-zero if metadata is invalid |
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*/ |
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static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) |
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{ |
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int metadata_type, metadata_size; |
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if (buf_size < FLAC_STREAMINFO_SIZE+8) { |
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/* need more data */ |
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return 0; |
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} |
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avpriv_flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size); |
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if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO || |
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metadata_size != FLAC_STREAMINFO_SIZE) { |
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return AVERROR_INVALIDDATA; |
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} |
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avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]); |
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allocate_buffers(s); |
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s->got_streaminfo = 1; |
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return 0; |
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} |
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/** |
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* Determine the size of an inline header. |
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* @param buf input buffer, starting with the "fLaC" marker |
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* @param buf_size buffer size |
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* @return number of bytes in the header, or 0 if more data is needed |
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*/ |
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static int get_metadata_size(const uint8_t *buf, int buf_size) |
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{ |
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int metadata_last, metadata_size; |
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const uint8_t *buf_end = buf + buf_size; |
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buf += 4; |
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do { |
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if (buf_end - buf < 4) |
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return 0; |
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avpriv_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size); |
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buf += 4; |
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if (buf_end - buf < metadata_size) { |
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/* need more data in order to read the complete header */ |
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return 0; |
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} |
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buf += metadata_size; |
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} while (!metadata_last); |
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return buf_size - (buf_end - buf); |
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} |
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static int decode_residuals(FLACContext *s, int channel, int pred_order) |
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{ |
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int i, tmp, partition, method_type, rice_order; |
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int sample = 0, samples; |
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method_type = get_bits(&s->gb, 2); |
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if (method_type > 1) { |
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av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n", |
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method_type); |
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return -1; |
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} |
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rice_order = get_bits(&s->gb, 4); |
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samples= s->blocksize >> rice_order; |
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if (pred_order > samples) { |
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av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", |
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pred_order, samples); |
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return -1; |
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} |
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sample= |
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i= pred_order; |
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for (partition = 0; partition < (1 << rice_order); partition++) { |
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tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5); |
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if (tmp == (method_type == 0 ? 15 : 31)) { |
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tmp = get_bits(&s->gb, 5); |
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for (; i < samples; i++, sample++) |
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s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp); |
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} else { |
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for (; i < samples; i++, sample++) { |
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s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0); |
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} |
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} |
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i= 0; |
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} |
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return 0; |
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} |
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static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order) |
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{ |
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const int blocksize = s->blocksize; |
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int32_t *decoded = s->decoded[channel]; |
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int a, b, c, d, i; |
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/* warm up samples */ |
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for (i = 0; i < pred_order; i++) { |
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decoded[i] = get_sbits_long(&s->gb, s->curr_bps); |
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} |
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if (decode_residuals(s, channel, pred_order) < 0) |
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return -1; |
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if (pred_order > 0) |
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a = decoded[pred_order-1]; |
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if (pred_order > 1) |
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b = a - decoded[pred_order-2]; |
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if (pred_order > 2) |
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c = b - decoded[pred_order-2] + decoded[pred_order-3]; |
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if (pred_order > 3) |
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d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4]; |
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switch (pred_order) { |
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case 0: |
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break; |
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case 1: |
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for (i = pred_order; i < blocksize; i++) |
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decoded[i] = a += decoded[i]; |
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break; |
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case 2: |
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for (i = pred_order; i < blocksize; i++) |
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decoded[i] = a += b += decoded[i]; |
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break; |
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case 3: |
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for (i = pred_order; i < blocksize; i++) |
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decoded[i] = a += b += c += decoded[i]; |
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break; |
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case 4: |
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for (i = pred_order; i < blocksize; i++) |
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decoded[i] = a += b += c += d += decoded[i]; |
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break; |
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default: |
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av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order); |
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return -1; |
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} |
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return 0; |
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} |
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static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) |
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{ |
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int i, j; |
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int coeff_prec, qlevel; |
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int coeffs[32]; |
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int32_t *decoded = s->decoded[channel]; |
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/* warm up samples */ |
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for (i = 0; i < pred_order; i++) { |
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decoded[i] = get_sbits_long(&s->gb, s->curr_bps); |
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} |
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coeff_prec = get_bits(&s->gb, 4) + 1; |
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if (coeff_prec == 16) { |
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av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n"); |
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return -1; |
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} |
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qlevel = get_sbits(&s->gb, 5); |
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if (qlevel < 0) { |
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av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n", |
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qlevel); |
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return -1; |
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} |
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for (i = 0; i < pred_order; i++) { |
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coeffs[i] = get_sbits(&s->gb, coeff_prec); |
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} |
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if (decode_residuals(s, channel, pred_order) < 0) |
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return -1; |
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if (s->bps > 16) { |
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int64_t sum; |
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for (i = pred_order; i < s->blocksize; i++) { |
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sum = 0; |
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for (j = 0; j < pred_order; j++) |
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sum += (int64_t)coeffs[j] * decoded[i-j-1]; |
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decoded[i] += sum >> qlevel; |
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} |
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} else { |
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for (i = pred_order; i < s->blocksize-1; i += 2) { |
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int c; |
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int d = decoded[i-pred_order]; |
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int s0 = 0, s1 = 0; |
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for (j = pred_order-1; j > 0; j--) { |
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c = coeffs[j]; |
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s0 += c*d; |
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d = decoded[i-j]; |
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s1 += c*d; |
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} |
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c = coeffs[0]; |
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s0 += c*d; |
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d = decoded[i] += s0 >> qlevel; |
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s1 += c*d; |
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decoded[i+1] += s1 >> qlevel; |
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} |
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if (i < s->blocksize) { |
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int sum = 0; |
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for (j = 0; j < pred_order; j++) |
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sum += coeffs[j] * decoded[i-j-1]; |
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decoded[i] += sum >> qlevel; |
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} |
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} |
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return 0; |
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} |
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static inline int decode_subframe(FLACContext *s, int channel) |
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{ |
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int type, wasted = 0; |
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int i, tmp; |
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s->curr_bps = s->bps; |
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if (channel == 0) { |
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if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE) |
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s->curr_bps++; |
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} else { |
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if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE) |
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s->curr_bps++; |
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} |
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if (get_bits1(&s->gb)) { |
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av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n"); |
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return -1; |
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} |
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type = get_bits(&s->gb, 6); |
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if (get_bits1(&s->gb)) { |
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wasted = 1; |
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while (!get_bits1(&s->gb)) |
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wasted++; |
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s->curr_bps -= wasted; |
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} |
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if (s->curr_bps > 32) { |
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av_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0); |
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return -1; |
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} |
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//FIXME use av_log2 for types |
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if (type == 0) { |
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tmp = get_sbits_long(&s->gb, s->curr_bps); |
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for (i = 0; i < s->blocksize; i++) |
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s->decoded[channel][i] = tmp; |
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} else if (type == 1) { |
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for (i = 0; i < s->blocksize; i++) |
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s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps); |
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} else if ((type >= 8) && (type <= 12)) { |
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if (decode_subframe_fixed(s, channel, type & ~0x8) < 0) |
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return -1; |
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} else if (type >= 32) { |
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if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0) |
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return -1; |
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} else { |
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av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n"); |
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return -1; |
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} |
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if (wasted) { |
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int i; |
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for (i = 0; i < s->blocksize; i++) |
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s->decoded[channel][i] <<= wasted; |
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} |
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return 0; |
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} |
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static int decode_frame(FLACContext *s) |
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{ |
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int i; |
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GetBitContext *gb = &s->gb; |
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FLACFrameInfo fi; |
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if (ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) { |
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av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n"); |
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return -1; |
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} |
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|
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if (s->channels && fi.channels != s->channels) { |
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av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream " |
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"is not supported\n"); |
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return -1; |
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} |
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s->channels = s->avctx->channels = fi.channels; |
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s->ch_mode = fi.ch_mode; |
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|
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if (!s->bps && !fi.bps) { |
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av_log(s->avctx, AV_LOG_ERROR, "bps not found in STREAMINFO or frame header\n"); |
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return -1; |
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} |
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if (!fi.bps) { |
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fi.bps = s->bps; |
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} else if (s->bps && fi.bps != s->bps) { |
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av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not " |
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"supported\n"); |
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return -1; |
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} |
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s->bps = s->avctx->bits_per_raw_sample = fi.bps; |
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|
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if (s->bps > 16) { |
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S32; |
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s->sample_shift = 32 - s->bps; |
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s->is32 = 1; |
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} else { |
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
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s->sample_shift = 16 - s->bps; |
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s->is32 = 0; |
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} |
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|
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if (!s->max_blocksize) |
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s->max_blocksize = FLAC_MAX_BLOCKSIZE; |
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if (fi.blocksize > s->max_blocksize) { |
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av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize, |
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s->max_blocksize); |
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return -1; |
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} |
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s->blocksize = fi.blocksize; |
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|
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if (!s->samplerate && !fi.samplerate) { |
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av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO" |
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" or frame header\n"); |
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return -1; |
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} |
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if (fi.samplerate == 0) { |
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fi.samplerate = s->samplerate; |
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} else if (s->samplerate && fi.samplerate != s->samplerate) { |
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av_log(s->avctx, AV_LOG_WARNING, "sample rate changed from %d to %d\n", |
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s->samplerate, fi.samplerate); |
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} |
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s->samplerate = s->avctx->sample_rate = fi.samplerate; |
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|
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if (!s->got_streaminfo) { |
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allocate_buffers(s); |
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s->got_streaminfo = 1; |
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dump_headers(s->avctx, (FLACStreaminfo *)s); |
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} |
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|
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// dump_headers(s->avctx, (FLACStreaminfo *)s); |
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|
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/* subframes */ |
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for (i = 0; i < s->channels; i++) { |
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if (decode_subframe(s, i) < 0) |
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return -1; |
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} |
|
|
|
align_get_bits(gb); |
|
|
|
/* frame footer */ |
|
skip_bits(gb, 16); /* data crc */ |
|
|
|
return 0; |
|
} |
|
|
|
static int flac_decode_frame(AVCodecContext *avctx, void *data, |
|
int *got_frame_ptr, AVPacket *avpkt) |
|
{ |
|
const uint8_t *buf = avpkt->data; |
|
int buf_size = avpkt->size; |
|
FLACContext *s = avctx->priv_data; |
|
int i, j = 0, bytes_read = 0; |
|
int16_t *samples_16; |
|
int32_t *samples_32; |
|
int ret; |
|
|
|
*got_frame_ptr = 0; |
|
|
|
if (s->max_framesize == 0) { |
|
s->max_framesize = |
|
ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE, |
|
FLAC_MAX_CHANNELS, 32); |
|
} |
|
|
|
/* check that there is at least the smallest decodable amount of data. |
|
this amount corresponds to the smallest valid FLAC frame possible. |
|
FF F8 69 02 00 00 9A 00 00 34 46 */ |
|
if (buf_size < FLAC_MIN_FRAME_SIZE) |
|
return buf_size; |
|
|
|
/* 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; |
|
} |
|
return get_metadata_size(buf, buf_size); |
|
} |
|
|
|
/* decode frame */ |
|
init_get_bits(&s->gb, buf, buf_size*8); |
|
if (decode_frame(s) < 0) { |
|
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n"); |
|
return -1; |
|
} |
|
bytes_read = (get_bits_count(&s->gb)+7)/8; |
|
|
|
/* get output buffer */ |
|
s->frame.nb_samples = s->blocksize; |
|
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { |
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); |
|
return ret; |
|
} |
|
samples_16 = (int16_t *)s->frame.data[0]; |
|
samples_32 = (int32_t *)s->frame.data[0]; |
|
|
|
#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->ch_mode) { |
|
case FLAC_CHMODE_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 FLAC_CHMODE_LEFT_SIDE: |
|
DECORRELATE(a,a-b) |
|
case FLAC_CHMODE_RIGHT_SIDE: |
|
DECORRELATE(a+b,b) |
|
case FLAC_CHMODE_MID_SIDE: |
|
DECORRELATE( (a-=b>>1) + b, a) |
|
} |
|
|
|
if (bytes_read > buf_size) { |
|
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size); |
|
return -1; |
|
} |
|
if (bytes_read < buf_size) { |
|
av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n", |
|
buf_size - bytes_read, buf_size); |
|
} |
|
|
|
*got_frame_ptr = 1; |
|
*(AVFrame *)data = s->frame; |
|
|
|
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]); |
|
} |
|
|
|
return 0; |
|
} |
|
|
|
AVCodec ff_flac_decoder = { |
|
.name = "flac", |
|
.type = AVMEDIA_TYPE_AUDIO, |
|
.id = CODEC_ID_FLAC, |
|
.priv_data_size = sizeof(FLACContext), |
|
.init = flac_decode_init, |
|
.close = flac_decode_close, |
|
.decode = flac_decode_frame, |
|
.capabilities = CODEC_CAP_DR1, |
|
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"), |
|
};
|
|
|