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https://github.com/FFmpeg/FFmpeg.git
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9d82b0dda1
Originally committed as revision 5664 to svn://svn.ffmpeg.org/ffmpeg/trunk
741 lines
21 KiB
C
741 lines
21 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 library 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 of the License, or (at your option) any later version.
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*
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* This library 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 this library; 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 flac.c
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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 "avcodec.h"
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#include "bitstream.h"
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#include "golomb.h"
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#include "crc.h"
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#undef NDEBUG
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#include <assert.h>
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#define MAX_CHANNELS 8
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#define MAX_BLOCKSIZE 65535
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#define FLAC_STREAMINFO_SIZE 34
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enum decorrelation_type {
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INDEPENDENT,
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LEFT_SIDE,
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RIGHT_SIDE,
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MID_SIDE,
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};
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typedef struct FLACContext {
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AVCodecContext *avctx;
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GetBitContext gb;
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int min_blocksize, max_blocksize;
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int min_framesize, max_framesize;
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int samplerate, channels;
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int blocksize/*, last_blocksize*/;
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int bps, curr_bps;
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enum decorrelation_type decorrelation;
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int32_t *decoded[MAX_CHANNELS];
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uint8_t *bitstream;
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int bitstream_size;
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int bitstream_index;
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unsigned int allocated_bitstream_size;
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} FLACContext;
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#define METADATA_TYPE_STREAMINFO 0
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static int sample_rate_table[] =
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{ 0, 0, 0, 0,
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8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
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0, 0, 0, 0 };
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static int sample_size_table[] =
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{ 0, 8, 12, 0, 16, 20, 24, 0 };
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static int blocksize_table[] = {
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0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0,
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256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
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};
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static int64_t get_utf8(GetBitContext *gb){
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int64_t val;
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GET_UTF8(val, get_bits(gb, 8), return -1;)
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return val;
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}
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static void metadata_streaminfo(FLACContext *s);
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static void dump_headers(FLACContext *s);
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static int flac_decode_init(AVCodecContext * avctx)
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{
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FLACContext *s = avctx->priv_data;
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s->avctx = avctx;
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/* initialize based on the demuxer-supplied streamdata header */
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if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
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init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
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metadata_streaminfo(s);
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dump_headers(s);
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}
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return 0;
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}
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static void dump_headers(FLACContext *s)
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{
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av_log(s->avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d (%d)\n", s->min_blocksize, s->max_blocksize, s->blocksize);
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av_log(s->avctx, AV_LOG_DEBUG, " Framesize: %d .. %d\n", s->min_framesize, s->max_framesize);
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av_log(s->avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate);
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av_log(s->avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels);
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av_log(s->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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int i;
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assert(s->max_blocksize);
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if(s->max_framesize == 0 && s->max_blocksize){
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s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; //FIXME header overhead
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}
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for (i = 0; i < s->channels; i++)
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{
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s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t)*s->max_blocksize);
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}
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s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
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}
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static void metadata_streaminfo(FLACContext *s)
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{
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/* mandatory streaminfo */
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s->min_blocksize = get_bits(&s->gb, 16);
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s->max_blocksize = get_bits(&s->gb, 16);
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s->min_framesize = get_bits_long(&s->gb, 24);
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s->max_framesize = get_bits_long(&s->gb, 24);
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s->samplerate = get_bits_long(&s->gb, 20);
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s->channels = get_bits(&s->gb, 3) + 1;
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s->bps = get_bits(&s->gb, 5) + 1;
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s->avctx->channels = s->channels;
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s->avctx->sample_rate = s->samplerate;
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skip_bits(&s->gb, 36); /* total num of samples */
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skip_bits(&s->gb, 64); /* md5 sum */
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skip_bits(&s->gb, 64); /* md5 sum */
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allocate_buffers(s);
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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 != 0){
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av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", 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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sample=
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i= pred_order;
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for (partition = 0; partition < (1 << rice_order); partition++)
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{
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tmp = get_bits(&s->gb, 4);
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if (tmp == 15)
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{
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av_log(s->avctx, AV_LOG_DEBUG, "fixed len partition\n");
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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(&s->gb, tmp);
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}
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else
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{
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// av_log(s->avctx, AV_LOG_DEBUG, "rice coded partition k=%d\n", tmp);
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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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// av_log(s->avctx, AV_LOG_DEBUG, "partitions: %d, samples: %d\n", 1 << rice_order, sample);
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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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int i;
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// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME FIXED\n");
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/* warm up samples */
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// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
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for (i = 0; i < pred_order; i++)
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{
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s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
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// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]);
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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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switch(pred_order)
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{
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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 < s->blocksize; i++)
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s->decoded[channel][i] += s->decoded[channel][i-1];
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break;
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case 2:
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for (i = pred_order; i < s->blocksize; i++)
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s->decoded[channel][i] += 2*s->decoded[channel][i-1]
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- s->decoded[channel][i-2];
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break;
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case 3:
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for (i = pred_order; i < s->blocksize; i++)
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s->decoded[channel][i] += 3*s->decoded[channel][i-1]
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- 3*s->decoded[channel][i-2]
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+ s->decoded[channel][i-3];
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break;
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case 4:
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for (i = pred_order; i < s->blocksize; i++)
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s->decoded[channel][i] += 4*s->decoded[channel][i-1]
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- 6*s->decoded[channel][i-2]
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+ 4*s->decoded[channel][i-3]
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- s->decoded[channel][i-4];
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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[pred_order];
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// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME LPC\n");
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/* warm up samples */
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// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
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for (i = 0; i < pred_order; i++)
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{
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s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
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// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]);
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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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{
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av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n");
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return -1;
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}
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// av_log(s->avctx, AV_LOG_DEBUG, " qlp coeff prec: %d\n", coeff_prec);
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qlevel = get_sbits(&s->gb, 5);
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// av_log(s->avctx, AV_LOG_DEBUG, " quant level: %d\n", qlevel);
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if(qlevel < 0){
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av_log(s->avctx, AV_LOG_DEBUG, "qlevel %d not supported, maybe buggy stream\n", 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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{
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coeffs[i] = get_sbits(&s->gb, coeff_prec);
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// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, coeffs[i]);
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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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{
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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] * s->decoded[channel][i-j-1];
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s->decoded[channel][i] += sum >> qlevel;
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}
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} else {
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int sum;
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for (i = pred_order; i < s->blocksize; i++)
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{
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sum = 0;
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for (j = 0; j < pred_order; j++)
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sum += coeffs[j] * s->decoded[channel][i-j-1];
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s->decoded[channel][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->decorrelation == RIGHT_SIDE)
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s->curr_bps++;
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}else{
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if(s->decorrelation == LEFT_SIDE || s->decorrelation == 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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{
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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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// wasted = get_bits1(&s->gb);
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// if (wasted)
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// {
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// while (!get_bits1(&s->gb))
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// wasted++;
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// if (wasted)
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// wasted++;
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// s->curr_bps -= wasted;
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// }
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#if 0
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wasted= 16 - av_log2(show_bits(&s->gb, 17));
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skip_bits(&s->gb, wasted+1);
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s->curr_bps -= wasted;
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#else
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if (get_bits1(&s->gb))
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{
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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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av_log(s->avctx, AV_LOG_DEBUG, "%d wasted bits\n", wasted);
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}
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#endif
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//FIXME use av_log2 for types
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if (type == 0)
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{
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av_log(s->avctx, AV_LOG_DEBUG, "coding type: constant\n");
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tmp = get_sbits(&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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}
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else if (type == 1)
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{
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av_log(s->avctx, AV_LOG_DEBUG, "coding type: verbatim\n");
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for (i = 0; i < s->blocksize; i++)
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s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
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}
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else if ((type >= 8) && (type <= 12))
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{
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// av_log(s->avctx, AV_LOG_DEBUG, "coding type: fixed\n");
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if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
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return -1;
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}
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else if (type >= 32)
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{
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// av_log(s->avctx, AV_LOG_DEBUG, "coding type: lpc\n");
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if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
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return -1;
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}
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else
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{
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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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{
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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 blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
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int decorrelation, bps, blocksize, samplerate;
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blocksize_code = get_bits(&s->gb, 4);
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sample_rate_code = get_bits(&s->gb, 4);
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assignment = get_bits(&s->gb, 4); /* channel assignment */
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if (assignment < 8 && s->channels == assignment+1)
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decorrelation = INDEPENDENT;
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else if (assignment >=8 && assignment < 11 && s->channels == 2)
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decorrelation = LEFT_SIDE + assignment - 8;
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else
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{
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av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels);
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return -1;
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}
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sample_size_code = get_bits(&s->gb, 3);
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if(sample_size_code == 0)
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bps= s->bps;
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else if((sample_size_code != 3) && (sample_size_code != 7))
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bps = sample_size_table[sample_size_code];
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else
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{
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av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", sample_size_code);
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return -1;
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}
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if (get_bits1(&s->gb))
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{
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av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
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return -1;
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}
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if(get_utf8(&s->gb) < 0){
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av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
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return -1;
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}
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#if 0
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if (/*((blocksize_code == 6) || (blocksize_code == 7)) &&*/
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(s->min_blocksize != s->max_blocksize)){
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}else{
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}
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#endif
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if (blocksize_code == 0)
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blocksize = s->min_blocksize;
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else if (blocksize_code == 6)
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blocksize = get_bits(&s->gb, 8)+1;
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else if (blocksize_code == 7)
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blocksize = get_bits(&s->gb, 16)+1;
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else
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blocksize = blocksize_table[blocksize_code];
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if(blocksize > s->max_blocksize){
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av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize, s->max_blocksize);
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return -1;
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}
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if (sample_rate_code == 0){
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samplerate= s->samplerate;
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}else if ((sample_rate_code > 3) && (sample_rate_code < 12))
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samplerate = sample_rate_table[sample_rate_code];
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else if (sample_rate_code == 12)
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samplerate = get_bits(&s->gb, 8) * 1000;
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else if (sample_rate_code == 13)
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samplerate = get_bits(&s->gb, 16);
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else if (sample_rate_code == 14)
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samplerate = get_bits(&s->gb, 16) * 10;
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else{
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av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n", sample_rate_code);
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return -1;
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}
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skip_bits(&s->gb, 8);
|
|
crc8= av_crc(av_crc07, 0, s->gb.buffer, get_bits_count(&s->gb)/8);
|
|
if(crc8){
|
|
av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
|
|
return -1;
|
|
}
|
|
|
|
s->blocksize = blocksize;
|
|
s->samplerate = samplerate;
|
|
s->bps = bps;
|
|
s->decorrelation= decorrelation;
|
|
|
|
// dump_headers(s);
|
|
|
|
/* subframes */
|
|
for (i = 0; i < s->channels; i++)
|
|
{
|
|
// av_log(s->avctx, AV_LOG_DEBUG, "decoded: %x residual: %x\n", s->decoded[i], s->residual[i]);
|
|
if (decode_subframe(s, i) < 0)
|
|
return -1;
|
|
}
|
|
|
|
align_get_bits(&s->gb);
|
|
|
|
/* frame footer */
|
|
skip_bits(&s->gb, 16); /* data crc */
|
|
|
|
return 0;
|
|
}
|
|
|
|
static inline int16_t shift_to_16_bits(int32_t data, int bps)
|
|
{
|
|
if (bps == 24) {
|
|
return (data >> 8);
|
|
} else if (bps == 20) {
|
|
return (data >> 4);
|
|
} else {
|
|
return data;
|
|
}
|
|
}
|
|
|
|
static int flac_decode_frame(AVCodecContext *avctx,
|
|
void *data, int *data_size,
|
|
uint8_t *buf, int buf_size)
|
|
{
|
|
FLACContext *s = avctx->priv_data;
|
|
int metadata_last, metadata_type, metadata_size;
|
|
int tmp = 0, i, j = 0, input_buf_size = 0;
|
|
int16_t *samples = data;
|
|
|
|
if(s->max_framesize == 0){
|
|
s->max_framesize= 65536; // should hopefully be enough for the first header
|
|
s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
|
|
}
|
|
|
|
if(1 && s->max_framesize){//FIXME truncated
|
|
buf_size= FFMAX(FFMIN(buf_size, s->max_framesize - s->bitstream_size), 0);
|
|
input_buf_size= buf_size;
|
|
|
|
if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
|
|
// printf("memmove\n");
|
|
memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
|
|
s->bitstream_index=0;
|
|
}
|
|
memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
|
|
buf= &s->bitstream[s->bitstream_index];
|
|
buf_size += s->bitstream_size;
|
|
s->bitstream_size= buf_size;
|
|
|
|
if(buf_size < s->max_framesize){
|
|
// printf("wanna more data ...\n");
|
|
return input_buf_size;
|
|
}
|
|
}
|
|
|
|
init_get_bits(&s->gb, buf, buf_size*8);
|
|
|
|
/* fLaC signature (be) */
|
|
if (show_bits_long(&s->gb, 32) == bswap_32(ff_get_fourcc("fLaC")))
|
|
{
|
|
skip_bits(&s->gb, 32);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n");
|
|
do {
|
|
metadata_last = get_bits(&s->gb, 1);
|
|
metadata_type = get_bits(&s->gb, 7);
|
|
metadata_size = get_bits_long(&s->gb, 24);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " metadata block: flag = %d, type = %d, size = %d\n",
|
|
metadata_last, metadata_type,
|
|
metadata_size);
|
|
if(metadata_size){
|
|
switch(metadata_type)
|
|
{
|
|
case METADATA_TYPE_STREAMINFO:{
|
|
metadata_streaminfo(s);
|
|
|
|
/* Buffer might have been reallocated, reinit bitreader */
|
|
if(buf != &s->bitstream[s->bitstream_index])
|
|
{
|
|
int bits_count = get_bits_count(&s->gb);
|
|
buf= &s->bitstream[s->bitstream_index];
|
|
init_get_bits(&s->gb, buf, buf_size*8);
|
|
skip_bits(&s->gb, bits_count);
|
|
}
|
|
|
|
dump_headers(s);
|
|
break;}
|
|
default:
|
|
for(i=0; i<metadata_size; i++)
|
|
skip_bits(&s->gb, 8);
|
|
}
|
|
}
|
|
} while(!metadata_last);
|
|
}
|
|
else
|
|
{
|
|
|
|
tmp = show_bits(&s->gb, 16);
|
|
if(tmp != 0xFFF8){
|
|
av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
|
|
while(get_bits_count(&s->gb)/8+2 < buf_size && show_bits(&s->gb, 16) != 0xFFF8)
|
|
skip_bits(&s->gb, 8);
|
|
goto end; // we may not have enough bits left to decode a frame, so try next time
|
|
}
|
|
skip_bits(&s->gb, 16);
|
|
if (decode_frame(s) < 0){
|
|
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
|
|
s->bitstream_size=0;
|
|
s->bitstream_index=0;
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
|
|
#if 0
|
|
/* fix the channel order here */
|
|
if (s->order == MID_SIDE)
|
|
{
|
|
short *left = samples;
|
|
short *right = samples + s->blocksize;
|
|
for (i = 0; i < s->blocksize; i += 2)
|
|
{
|
|
uint32_t x = s->decoded[0][i];
|
|
uint32_t y = s->decoded[0][i+1];
|
|
|
|
right[i] = x - (y / 2);
|
|
left[i] = right[i] + y;
|
|
}
|
|
*data_size = 2 * s->blocksize;
|
|
}
|
|
else
|
|
{
|
|
for (i = 0; i < s->channels; i++)
|
|
{
|
|
switch(s->order)
|
|
{
|
|
case INDEPENDENT:
|
|
for (j = 0; j < s->blocksize; j++)
|
|
samples[(s->blocksize*i)+j] = s->decoded[i][j];
|
|
break;
|
|
case LEFT_SIDE:
|
|
case RIGHT_SIDE:
|
|
if (i == 0)
|
|
for (j = 0; j < s->blocksize; j++)
|
|
samples[(s->blocksize*i)+j] = s->decoded[0][j];
|
|
else
|
|
for (j = 0; j < s->blocksize; j++)
|
|
samples[(s->blocksize*i)+j] = s->decoded[0][j] - s->decoded[i][j];
|
|
break;
|
|
// case MID_SIDE:
|
|
// av_log(s->avctx, AV_LOG_DEBUG, "mid-side unsupported\n");
|
|
}
|
|
*data_size += s->blocksize;
|
|
}
|
|
}
|
|
#else
|
|
#define DECORRELATE(left, right)\
|
|
assert(s->channels == 2);\
|
|
for (i = 0; i < s->blocksize; i++)\
|
|
{\
|
|
int a= s->decoded[0][i];\
|
|
int b= s->decoded[1][i];\
|
|
*(samples++) = (left ) >> (16 - s->bps);\
|
|
*(samples++) = (right) >> (16 - s->bps);\
|
|
}\
|
|
break;
|
|
|
|
switch(s->decorrelation)
|
|
{
|
|
case INDEPENDENT:
|
|
for (j = 0; j < s->blocksize; j++)
|
|
{
|
|
for (i = 0; i < s->channels; i++)
|
|
*(samples++) = shift_to_16_bits(s->decoded[i][j], s->bps);
|
|
}
|
|
break;
|
|
case LEFT_SIDE:
|
|
DECORRELATE(a,a-b)
|
|
case RIGHT_SIDE:
|
|
DECORRELATE(a+b,b)
|
|
case MID_SIDE:
|
|
DECORRELATE( (a-=b>>1) + b, a)
|
|
}
|
|
#endif
|
|
|
|
*data_size = (int8_t *)samples - (int8_t *)data;
|
|
// av_log(s->avctx, AV_LOG_DEBUG, "data size: %d\n", *data_size);
|
|
|
|
// s->last_blocksize = s->blocksize;
|
|
end:
|
|
i= (get_bits_count(&s->gb)+7)/8;;
|
|
if(i > buf_size){
|
|
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
|
|
s->bitstream_size=0;
|
|
s->bitstream_index=0;
|
|
return -1;
|
|
}
|
|
|
|
if(s->bitstream_size){
|
|
s->bitstream_index += i;
|
|
s->bitstream_size -= i;
|
|
return input_buf_size;
|
|
}else
|
|
return i;
|
|
}
|
|
|
|
static int flac_decode_close(AVCodecContext *avctx)
|
|
{
|
|
FLACContext *s = avctx->priv_data;
|
|
int i;
|
|
|
|
for (i = 0; i < s->channels; i++)
|
|
{
|
|
av_freep(&s->decoded[i]);
|
|
}
|
|
av_freep(&s->bitstream);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void flac_flush(AVCodecContext *avctx){
|
|
FLACContext *s = avctx->priv_data;
|
|
|
|
s->bitstream_size=
|
|
s->bitstream_index= 0;
|
|
}
|
|
|
|
AVCodec flac_decoder = {
|
|
"flac",
|
|
CODEC_TYPE_AUDIO,
|
|
CODEC_ID_FLAC,
|
|
sizeof(FLACContext),
|
|
flac_decode_init,
|
|
NULL,
|
|
flac_decode_close,
|
|
flac_decode_frame,
|
|
.flush= flac_flush,
|
|
};
|