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680 lines
22 KiB
C
Executable File
680 lines
22 KiB
C
Executable File
/*
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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/avassert.h"
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#include "libavutil/crc.h"
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#include "libavutil/opt.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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#include "thread.h"
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#include "unary.h"
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typedef struct FLACContext {
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AVClass *class;
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struct FLACStreaminfo flac_stream_info;
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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 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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uint8_t *decoded_buffer;
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unsigned int decoded_buffer_size;
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int buggy_lpc; ///< use workaround for old lavc encoded files
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FLACDSPContext dsp;
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} FLACContext;
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static int 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->flac_stream_info.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->flac_stream_info.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->flac_stream_info.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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int ret;
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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 (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo))
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return AVERROR_INVALIDDATA;
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/* initialize based on the demuxer-supplied streamdata header */
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ret = ff_flac_parse_streaminfo(avctx, &s->flac_stream_info, streaminfo);
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if (ret < 0)
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return ret;
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ret = allocate_buffers(s);
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if (ret < 0)
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return ret;
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flac_set_bps(s);
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ff_flacdsp_init(&s->dsp, avctx->sample_fmt,
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s->flac_stream_info.channels, s->flac_stream_info.bps);
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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 int allocate_buffers(FLACContext *s)
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{
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int buf_size;
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int ret;
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av_assert0(s->flac_stream_info.max_blocksize);
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buf_size = av_samples_get_buffer_size(NULL, s->flac_stream_info.channels,
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s->flac_stream_info.max_blocksize,
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AV_SAMPLE_FMT_S32P, 0);
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if (buf_size < 0)
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return buf_size;
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av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size);
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if (!s->decoded_buffer)
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return AVERROR(ENOMEM);
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ret = av_samples_fill_arrays((uint8_t **)s->decoded, NULL,
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s->decoded_buffer,
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s->flac_stream_info.channels,
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s->flac_stream_info.max_blocksize,
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AV_SAMPLE_FMT_S32P, 0);
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return ret < 0 ? ret : 0;
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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, ret;
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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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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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ret = ff_flac_parse_streaminfo(s->avctx, &s->flac_stream_info, &buf[8]);
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if (ret < 0)
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return ret;
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ret = allocate_buffers(s);
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if (ret < 0)
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return ret;
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flac_set_bps(s);
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ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt,
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s->flac_stream_info.channels, s->flac_stream_info.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 AVERROR_INVALIDDATA;
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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 AVERROR_INVALIDDATA;
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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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GetBitContext gb = s->gb;
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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(&gb, 2);
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rice_order = get_bits(&gb, 4);
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samples = s->blocksize >> rice_order;
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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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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 AVERROR_INVALIDDATA;
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}
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if (samples << rice_order != s->blocksize) {
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av_log(s->avctx, AV_LOG_ERROR, "invalid rice order: %i blocksize %i\n",
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rice_order, s->blocksize);
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return AVERROR_INVALIDDATA;
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}
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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 AVERROR_INVALIDDATA;
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}
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for (partition = 0; partition < (1 << rice_order); partition++) {
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tmp = get_bits(&gb, rice_bits);
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if (tmp == rice_esc) {
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tmp = get_bits(&gb, 5);
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for (; i < samples; i++)
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*decoded++ = get_sbits_long(&gb, tmp);
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} else {
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int real_limit = tmp ? (INT_MAX >> tmp) + 2 : INT_MAX;
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for (; i < samples; i++) {
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int v = get_sr_golomb_flac(&gb, tmp, real_limit, 0);
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if (v == 0x80000000){
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av_log(s->avctx, AV_LOG_ERROR, "invalid residual\n");
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return AVERROR_INVALIDDATA;
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}
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*decoded++ = v;
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}
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}
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i= 0;
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}
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s->gb = gb;
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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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unsigned av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d);
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int i;
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int ret;
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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 ((ret = decode_residuals(s, decoded, pred_order)) < 0)
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return ret;
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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] + 2U*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 AVERROR_INVALIDDATA;
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}
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return 0;
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}
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static void lpc_analyze_remodulate(SUINT32 *decoded, const int coeffs[32],
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int order, int qlevel, int len, int bps)
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{
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int i, j;
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int ebps = 1 << (bps-1);
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unsigned sigma = 0;
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for (i = order; i < len; i++)
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sigma |= decoded[i] + ebps;
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if (sigma < 2*ebps)
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return;
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for (i = len - 1; i >= order; i--) {
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int64_t p = 0;
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for (j = 0; j < order; j++)
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p += coeffs[j] * (int64_t)(int32_t)decoded[i-order+j];
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decoded[i] -= p >> qlevel;
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}
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for (i = order; i < len; i++, decoded++) {
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int32_t p = 0;
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for (j = 0; j < order; j++)
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p += coeffs[j] * (uint32_t)decoded[j];
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decoded[j] += p >> qlevel;
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}
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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, ret;
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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 AVERROR_INVALIDDATA;
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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 AVERROR_INVALIDDATA;
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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 ((ret = decode_residuals(s, decoded, pred_order)) < 0)
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return ret;
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if ( ( s->buggy_lpc && s->flac_stream_info.bps <= 16)
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|| ( !s->buggy_lpc && bps <= 16
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&& bps + coeff_prec + av_log2(pred_order) <= 32)) {
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s->dsp.lpc16(decoded, coeffs, pred_order, qlevel, s->blocksize);
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} else {
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s->dsp.lpc32(decoded, coeffs, pred_order, qlevel, s->blocksize);
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if (s->flac_stream_info.bps <= 16)
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lpc_analyze_remodulate(decoded, coeffs, pred_order, qlevel, s->blocksize, bps);
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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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int32_t *decoded = s->decoded[channel];
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int type, wasted = 0;
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int bps = s->flac_stream_info.bps;
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int i, tmp, ret;
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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 AVERROR_INVALIDDATA;
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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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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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wasted = 1 + get_unary(&s->gb, 1, get_bits_left(&s->gb));
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bps -= wasted;
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}
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if (bps > 32) {
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avpriv_report_missing_feature(s->avctx, "Decorrelated bit depth > 32");
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return AVERROR_PATCHWELCOME;
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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 ((ret = decode_subframe_fixed(s, decoded, type & ~0x8, bps)) < 0)
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return ret;
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} else if (type >= 32) {
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if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0)
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return ret;
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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 AVERROR_INVALIDDATA;
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}
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if (wasted && wasted < 32) {
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int i;
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for (i = 0; i < s->blocksize; i++)
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decoded[i] = (unsigned)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, ret;
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GetBitContext *gb = &s->gb;
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FLACFrameInfo fi;
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if ((ret = ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n");
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return ret;
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}
|
|
|
|
if ( s->flac_stream_info.channels
|
|
&& fi.channels != s->flac_stream_info.channels
|
|
&& s->got_streaminfo) {
|
|
s->flac_stream_info.channels = s->avctx->channels = fi.channels;
|
|
ff_flac_set_channel_layout(s->avctx);
|
|
ret = allocate_buffers(s);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
s->flac_stream_info.channels = s->avctx->channels = fi.channels;
|
|
if (!s->avctx->channel_layout)
|
|
ff_flac_set_channel_layout(s->avctx);
|
|
s->ch_mode = fi.ch_mode;
|
|
|
|
if (!s->flac_stream_info.bps && !fi.bps) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "bps not found in STREAMINFO or frame header\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (!fi.bps) {
|
|
fi.bps = s->flac_stream_info.bps;
|
|
} else if (s->flac_stream_info.bps && fi.bps != s->flac_stream_info.bps) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
|
|
"supported\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (!s->flac_stream_info.bps) {
|
|
s->flac_stream_info.bps = s->avctx->bits_per_raw_sample = fi.bps;
|
|
flac_set_bps(s);
|
|
}
|
|
|
|
if (!s->flac_stream_info.max_blocksize)
|
|
s->flac_stream_info.max_blocksize = FLAC_MAX_BLOCKSIZE;
|
|
if (fi.blocksize > s->flac_stream_info.max_blocksize) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize,
|
|
s->flac_stream_info.max_blocksize);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
s->blocksize = fi.blocksize;
|
|
|
|
if (!s->flac_stream_info.samplerate && !fi.samplerate) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO"
|
|
" or frame header\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (fi.samplerate == 0)
|
|
fi.samplerate = s->flac_stream_info.samplerate;
|
|
s->flac_stream_info.samplerate = s->avctx->sample_rate = fi.samplerate;
|
|
|
|
if (!s->got_streaminfo) {
|
|
ret = allocate_buffers(s);
|
|
if (ret < 0)
|
|
return ret;
|
|
s->got_streaminfo = 1;
|
|
dump_headers(s->avctx, &s->flac_stream_info);
|
|
}
|
|
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt,
|
|
s->flac_stream_info.channels, s->flac_stream_info.bps);
|
|
|
|
// dump_headers(s->avctx, &s->flac_stream_info);
|
|
|
|
/* subframes */
|
|
for (i = 0; i < s->flac_stream_info.channels; i++) {
|
|
if ((ret = decode_subframe(s, i)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
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)
|
|
{
|
|
AVFrame *frame = data;
|
|
ThreadFrame tframe = { .f = data };
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
FLACContext *s = avctx->priv_data;
|
|
int bytes_read = 0;
|
|
int ret;
|
|
|
|
*got_frame_ptr = 0;
|
|
|
|
if (s->flac_stream_info.max_framesize == 0) {
|
|
s->flac_stream_info.max_framesize =
|
|
ff_flac_get_max_frame_size(s->flac_stream_info.max_blocksize ? s->flac_stream_info.max_blocksize : FLAC_MAX_BLOCKSIZE,
|
|
FLAC_MAX_CHANNELS, 32);
|
|
}
|
|
|
|
if (buf_size > 5 && !memcmp(buf, "\177FLAC", 5)) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "skipping flac header packet 1\n");
|
|
return buf_size;
|
|
}
|
|
|
|
if (buf_size > 0 && (*buf & 0x7F) == FLAC_METADATA_TYPE_VORBIS_COMMENT) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "skipping vorbis comment\n");
|
|
return buf_size;
|
|
}
|
|
|
|
/* 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 && (ret = parse_streaminfo(s, buf, buf_size))) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
|
|
return ret;
|
|
}
|
|
return get_metadata_size(buf, buf_size);
|
|
}
|
|
|
|
/* decode frame */
|
|
if ((ret = init_get_bits8(&s->gb, buf, buf_size)) < 0)
|
|
return ret;
|
|
if ((ret = decode_frame(s)) < 0) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
|
|
return ret;
|
|
}
|
|
bytes_read = get_bits_count(&s->gb)/8;
|
|
|
|
if ((s->avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_COMPLIANT)) &&
|
|
av_crc(av_crc_get_table(AV_CRC_16_ANSI),
|
|
0, buf, bytes_read)) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "CRC error at PTS %"PRId64"\n", avpkt->pts);
|
|
if (s->avctx->err_recognition & AV_EF_EXPLODE)
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
/* get output buffer */
|
|
frame->nb_samples = s->blocksize;
|
|
if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0)
|
|
return ret;
|
|
|
|
s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded,
|
|
s->flac_stream_info.channels,
|
|
s->blocksize, s->sample_shift);
|
|
|
|
if (bytes_read > buf_size) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
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;
|
|
|
|
return bytes_read;
|
|
}
|
|
|
|
static av_cold int flac_decode_close(AVCodecContext *avctx)
|
|
{
|
|
FLACContext *s = avctx->priv_data;
|
|
|
|
av_freep(&s->decoded_buffer);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static const AVOption options[] = {
|
|
{ "use_buggy_lpc", "emulate old buggy lavc behavior", offsetof(FLACContext, buggy_lpc), AV_OPT_TYPE_BOOL, {.i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
|
|
{ NULL },
|
|
};
|
|
|
|
static const AVClass flac_decoder_class = {
|
|
"FLAC decoder",
|
|
av_default_item_name,
|
|
options,
|
|
LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVCodec ff_flac_decoder = {
|
|
.name = "flac",
|
|
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
|
|
.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 = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_FRAME_THREADS,
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
|
|
AV_SAMPLE_FMT_S16P,
|
|
AV_SAMPLE_FMT_S32,
|
|
AV_SAMPLE_FMT_S32P,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.priv_class = &flac_decoder_class,
|
|
};
|