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50df56161f
* qatar/master: dirac: use meaningful return values flacdec: simplify sample buffer handling flacdec: simplify loop in decode_residuals() fate: make Ut Video encoder tests use bitexact swscale flags build: amrwb: Drop redundant lsp dependency declaration fate: fix utvideoenc tests Conflicts: libavcodec/dirac.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
578 lines
17 KiB
C
578 lines
17 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 FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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* @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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#include <limits.h>
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#include "libavutil/audioconvert.h"
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#include "libavutil/avassert.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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#include "flacdsp.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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AVFrame frame;
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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 sample_shift; ///< shift required to make output samples 16-bit or 32-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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FLACDSPContext dsp;
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} FLACContext;
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static const int64_t flac_channel_layouts[6] = {
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AV_CH_LAYOUT_MONO,
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AV_CH_LAYOUT_STEREO,
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AV_CH_LAYOUT_SURROUND,
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AV_CH_LAYOUT_QUAD,
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AV_CH_LAYOUT_5POINT0,
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AV_CH_LAYOUT_5POINT1
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};
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static void allocate_buffers(FLACContext *s);
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static void flac_set_bps(FLACContext *s)
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{
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enum AVSampleFormat req = s->avctx->request_sample_fmt;
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int need32 = s->bps > 16;
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int want32 = av_get_bytes_per_sample(req) > 2;
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int planar = av_sample_fmt_is_planar(req);
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if (need32 || want32) {
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if (planar)
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
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else
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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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} else {
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if (planar)
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
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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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}
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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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/* 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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/* initialize based on the demuxer-supplied streamdata header */
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avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
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allocate_buffers(s);
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flac_set_bps(s);
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ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps);
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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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if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts))
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avctx->channel_layout = flac_channel_layouts[avctx->channels - 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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av_assert0(s->max_blocksize);
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for (i = 0; i < s->channels; i++) {
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s->decoded[i] = av_malloc(sizeof(int32_t)*s->max_blocksize);
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}
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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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flac_set_bps(s);
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ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
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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, int32_t *decoded, int pred_order)
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{
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int i, tmp, partition, method_type, rice_order;
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int rice_bits, rice_esc;
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int 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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rice_bits = 4 + method_type;
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rice_esc = (1 << rice_bits) - 1;
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decoded += pred_order;
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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, rice_bits);
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if (tmp == rice_esc) {
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tmp = get_bits(&s->gb, 5);
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for (; i < samples; i++)
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*decoded++ = get_sbits_long(&s->gb, tmp);
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} else {
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for (; i < samples; i++) {
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*decoded++ = 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, int32_t *decoded,
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int pred_order, int bps)
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{
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const int blocksize = s->blocksize;
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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, bps);
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}
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if (decode_residuals(s, decoded, 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, int32_t *decoded, int pred_order,
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int bps)
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{
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int i;
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int coeff_prec, qlevel;
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int coeffs[32];
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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, 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[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec);
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}
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if (decode_residuals(s, decoded, pred_order) < 0)
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return -1;
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s->dsp.lpc(decoded, coeffs, pred_order, qlevel, s->blocksize);
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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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int32_t *decoded = s->decoded[channel];
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int type, wasted = 0;
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int bps = s->bps;
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int i, tmp;
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if (channel == 0) {
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if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
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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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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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int left = get_bits_left(&s->gb);
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wasted = 1;
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if ( left < 0 ||
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(left < bps && !show_bits_long(&s->gb, left)) ||
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!show_bits_long(&s->gb, bps)) {
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av_log(s->avctx, AV_LOG_ERROR,
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"Invalid number of wasted bits > available bits (%d) - left=%d\n",
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bps, left);
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return AVERROR_INVALIDDATA;
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}
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while (!get_bits1(&s->gb))
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wasted++;
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bps -= wasted;
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}
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if (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, bps);
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for (i = 0; i < s->blocksize; i++)
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decoded[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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decoded[i] = get_sbits_long(&s->gb, bps);
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} else if ((type >= 8) && (type <= 12)) {
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if (decode_subframe_fixed(s, decoded, type & ~0x8, bps) < 0)
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return -1;
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} else if (type >= 32) {
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if (decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps) < 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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decoded[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) {
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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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if (!s->bps) {
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s->bps = s->avctx->bits_per_raw_sample = fi.bps;
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flac_set_bps(s);
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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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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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if (!s->got_streaminfo) {
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allocate_buffers(s);
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ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
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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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// dump_headers(s->avctx, (FLACStreaminfo *)s);
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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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}
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align_get_bits(gb);
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/* frame footer */
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skip_bits(gb, 16); /* data crc */
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return 0;
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}
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static int flac_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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FLACContext *s = avctx->priv_data;
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int bytes_read = 0;
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int ret;
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*got_frame_ptr = 0;
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|
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if (s->max_framesize == 0) {
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s->max_framesize =
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ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE,
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FLAC_MAX_CHANNELS, 32);
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}
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/* check that there is at least the smallest decodable amount of data.
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this amount corresponds to the smallest valid FLAC frame possible.
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FF F8 69 02 00 00 9A 00 00 34 46 */
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if (buf_size < FLAC_MIN_FRAME_SIZE)
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return buf_size;
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|
|
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/* check for inline header */
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if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
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if (!s->got_streaminfo && parse_streaminfo(s, buf, buf_size)) {
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av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
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return -1;
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}
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return get_metadata_size(buf, buf_size);
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}
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|
|
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/* decode frame */
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init_get_bits(&s->gb, buf, buf_size*8);
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if (decode_frame(s) < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
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return -1;
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}
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bytes_read = (get_bits_count(&s->gb)+7)/8;
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|
|
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/* get output buffer */
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s->frame.nb_samples = s->blocksize;
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if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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|
|
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s->dsp.decorrelate[s->ch_mode](s->frame.data, s->decoded, s->channels,
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s->blocksize, s->sample_shift);
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|
|
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if (bytes_read > buf_size) {
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av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
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return -1;
|
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}
|
|
if (bytes_read < buf_size) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n",
|
|
buf_size - bytes_read, buf_size);
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}
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|
|
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*got_frame_ptr = 1;
|
|
*(AVFrame *)data = s->frame;
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|
|
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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++) {
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|
av_freep(&s->decoded[i]);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_flac_decoder = {
|
|
.name = "flac",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_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)"),
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
|
|
AV_SAMPLE_FMT_S16P,
|
|
AV_SAMPLE_FMT_S32,
|
|
AV_SAMPLE_FMT_S32P,
|
|
-1 },
|
|
};
|