mirror of
https://github.com/xenia-project/FFmpeg.git
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2912e87a6c
Signed-off-by: Mans Rullgard <mans@mansr.com>
667 lines
20 KiB
C
667 lines
20 KiB
C
/*
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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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* @file
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* FLAC (Free Lossless Audio Codec) decoder
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* @author Alex Beregszaszi
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*
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* For more information on the FLAC format, visit:
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* 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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#include <limits.h>
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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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#undef NDEBUG
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#include <assert.h>
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typedef struct FLACContext {
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FLACSTREAMINFO
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AVCodecContext *avctx; ///< parent AVCodecContext
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GetBitContext gb; ///< GetBitContext initialized to start at the current frame
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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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static void allocate_buffers(FLACContext *s);
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int ff_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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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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/* 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 (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo))
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return -1;
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/* initialize based on the demuxer-supplied streamdata header */
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ff_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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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 ff_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 ff_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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ff_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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ff_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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ff_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
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buf += 4;
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if (buf + metadata_size > buf_end) {
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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 av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(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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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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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) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
|
|
"supported\n");
|
|
return -1;
|
|
}
|
|
s->bps = s->avctx->bits_per_raw_sample = fi.bps;
|
|
|
|
if (s->bps > 16) {
|
|
s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
|
|
s->sample_shift = 32 - s->bps;
|
|
s->is32 = 1;
|
|
} else {
|
|
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
|
|
s->sample_shift = 16 - s->bps;
|
|
s->is32 = 0;
|
|
}
|
|
|
|
if (!s->max_blocksize)
|
|
s->max_blocksize = FLAC_MAX_BLOCKSIZE;
|
|
if (fi.blocksize > s->max_blocksize) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize,
|
|
s->max_blocksize);
|
|
return -1;
|
|
}
|
|
s->blocksize = fi.blocksize;
|
|
|
|
if (!s->samplerate && !fi.samplerate) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO"
|
|
" or frame header\n");
|
|
return -1;
|
|
}
|
|
if (fi.samplerate == 0) {
|
|
fi.samplerate = s->samplerate;
|
|
} else if (s->samplerate && fi.samplerate != s->samplerate) {
|
|
av_log(s->avctx, AV_LOG_WARNING, "sample rate changed from %d to %d\n",
|
|
s->samplerate, fi.samplerate);
|
|
}
|
|
s->samplerate = s->avctx->sample_rate = fi.samplerate;
|
|
|
|
if (!s->got_streaminfo) {
|
|
allocate_buffers(s);
|
|
s->got_streaminfo = 1;
|
|
dump_headers(s->avctx, (FLACStreaminfo *)s);
|
|
}
|
|
|
|
// 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(gb);
|
|
|
|
/* frame footer */
|
|
skip_bits(gb, 16); /* data crc */
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int flac_decode_frame(AVCodecContext *avctx,
|
|
void *data, int *data_size,
|
|
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 = data;
|
|
int32_t *samples_32 = data;
|
|
int alloc_data_size= *data_size;
|
|
int output_size;
|
|
|
|
*data_size=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;
|
|
|
|
/* check if allocated data size is large enough for output */
|
|
output_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
|
|
if (output_size > alloc_data_size) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
|
|
"allocated data size\n");
|
|
return -1;
|
|
}
|
|
*data_size = output_size;
|
|
|
|
#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);
|
|
}
|
|
|
|
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 = {
|
|
"flac",
|
|
AVMEDIA_TYPE_AUDIO,
|
|
CODEC_ID_FLAC,
|
|
sizeof(FLACContext),
|
|
flac_decode_init,
|
|
NULL,
|
|
flac_decode_close,
|
|
flac_decode_frame,
|
|
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
|
|
};
|