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https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
share sample rate and blocksize tables between the FLAC encoder and FLAC
decoder Originally committed as revision 18089 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -83,8 +83,8 @@ OBJS-$(CONFIG_FFV1_DECODER) += ffv1.o rangecoder.o
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OBJS-$(CONFIG_FFV1_ENCODER) += ffv1.o rangecoder.o
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OBJS-$(CONFIG_FFVHUFF_DECODER) += huffyuv.o
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OBJS-$(CONFIG_FFVHUFF_ENCODER) += huffyuv.o
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OBJS-$(CONFIG_FLAC_DECODER) += flacdec.o
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OBJS-$(CONFIG_FLAC_ENCODER) += flacenc.o lpc.o
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OBJS-$(CONFIG_FLAC_DECODER) += flacdec.o flacdata.o
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OBJS-$(CONFIG_FLAC_ENCODER) += flacenc.o flacdata.o lpc.o
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OBJS-$(CONFIG_FLASHSV_DECODER) += flashsv.o
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OBJS-$(CONFIG_FLASHSV_ENCODER) += flashsvenc.o
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OBJS-$(CONFIG_FLIC_DECODER) += flicvideo.o
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@ -346,17 +346,17 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcm.o
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# libavformat dependencies
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OBJS-$(CONFIG_EAC3_DEMUXER) += ac3_parser.o ac3tab.o aac_ac3_parser.o
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OBJS-$(CONFIG_FLAC_DEMUXER) += flacdec.o
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OBJS-$(CONFIG_FLAC_MUXER) += flacdec.o
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OBJS-$(CONFIG_FLAC_DEMUXER) += flacdec.o flacdata.o
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OBJS-$(CONFIG_FLAC_MUXER) += flacdec.o flacdata.o
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OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
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OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o flacdec.o
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OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o flacdec.o flacdata.o
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OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o
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OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o flacdec.o
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OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o flacdec.o flacdata.o
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OBJS-$(CONFIG_MOV_DEMUXER) += mpeg4audio.o mpegaudiodata.o
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OBJS-$(CONFIG_MPEGTS_MUXER) += mpegvideo.o
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OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
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OBJS-$(CONFIG_OGG_DEMUXER) += flacdec.o
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OBJS-$(CONFIG_OGG_MUXER) += xiph.o flacdec.o
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OBJS-$(CONFIG_OGG_DEMUXER) += flacdec.o flacdata.o
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OBJS-$(CONFIG_OGG_MUXER) += xiph.o flacdec.o flacdata.o
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OBJS-$(CONFIG_RTP_MUXER) += mpegvideo.o
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# external codec libraries
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33
libavcodec/flacdata.c
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33
libavcodec/flacdata.c
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@ -0,0 +1,33 @@
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/*
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* FLAC data
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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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#include "internal.h"
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const int ff_flac_sample_rate_table[16] =
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{ 0,
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88200, 176400, 192000,
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8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
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0, 0, 0, 0 };
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const int16_t ff_flac_blocksize_table[16] = {
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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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31
libavcodec/flacdata.h
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31
libavcodec/flacdata.h
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@ -0,0 +1,31 @@
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/*
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* FLAC data header
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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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#ifndef AVCODEC_FLACDATA_H
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#define AVCODEC_FLACDATA_H
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#include "internal.h"
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extern const int ff_flac_sample_rate_table[16];
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extern const int16_t ff_flac_blocksize_table[16];
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#endif /* AVCODEC_FLACDATA_H */
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@ -42,6 +42,7 @@
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#include "bytestream.h"
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#include "golomb.h"
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#include "flac.h"
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#include "flacdata.h"
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#undef NDEBUG
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#include <assert.h>
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@ -66,20 +67,9 @@ typedef struct FLACContext {
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unsigned int allocated_bitstream_size;
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} FLACContext;
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static const int sample_rate_table[] =
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{ 0,
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88200, 176400, 192000,
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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 const int sample_size_table[] =
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{ 0, 8, 12, 0, 16, 20, 24, 0 };
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static const 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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{
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int64_t val;
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@ -547,7 +537,7 @@ static int decode_frame(FLACContext *s, int alloc_data_size)
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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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blocksize = ff_flac_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,
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@ -561,7 +551,7 @@ static int decode_frame(FLACContext *s, int alloc_data_size)
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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 < 12)
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samplerate = sample_rate_table[sample_rate_code];
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samplerate = ff_flac_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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@ -28,6 +28,7 @@
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#include "golomb.h"
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#include "lpc.h"
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#include "flac.h"
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#include "flacdata.h"
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#define FLAC_SUBFRAME_CONSTANT 0
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#define FLAC_SUBFRAME_VERBATIM 1
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@ -79,12 +80,10 @@ typedef struct FlacFrame {
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} FlacFrame;
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typedef struct FlacEncodeContext {
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FLACSTREAMINFO
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PutBitContext pb;
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int channels;
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int samplerate;
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int sr_code[2];
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int min_framesize;
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int max_framesize;
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int max_encoded_framesize;
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uint32_t frame_count;
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uint64_t sample_count;
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@ -96,20 +95,6 @@ typedef struct FlacEncodeContext {
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struct AVMD5 *md5ctx;
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} FlacEncodeContext;
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static const int flac_samplerates[16] = {
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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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};
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static const int flac_blocksizes[16] = {
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0,
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192,
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576, 1152, 2304, 4608,
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0, 0,
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256, 512, 1024, 2048, 4096, 8192, 16384, 32768
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};
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/**
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* Writes streaminfo metadata block to byte array
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*/
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@ -146,11 +131,11 @@ static int select_blocksize(int samplerate, int block_time_ms)
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int blocksize;
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assert(samplerate > 0);
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blocksize = flac_blocksizes[1];
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blocksize = ff_flac_blocksize_table[1];
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target = (samplerate * block_time_ms) / 1000;
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for(i=0; i<16; i++) {
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if(target >= flac_blocksizes[i] && flac_blocksizes[i] > blocksize) {
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blocksize = flac_blocksizes[i];
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if(target >= ff_flac_blocksize_table[i] && ff_flac_blocksize_table[i] > blocksize) {
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blocksize = ff_flac_blocksize_table[i];
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}
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}
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return blocksize;
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@ -181,8 +166,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
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if(freq < 1)
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return -1;
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for(i=4; i<12; i++) {
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if(freq == flac_samplerates[i]) {
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s->samplerate = flac_samplerates[i];
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if(freq == ff_flac_sample_rate_table[i]) {
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s->samplerate = ff_flac_sample_rate_table[i];
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s->sr_code[0] = i;
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s->sr_code[1] = 0;
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break;
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@ -392,8 +377,8 @@ static void init_frame(FlacEncodeContext *s)
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frame = &s->frame;
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for(i=0; i<16; i++) {
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if(s->avctx->frame_size == flac_blocksizes[i]) {
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frame->blocksize = flac_blocksizes[i];
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if(s->avctx->frame_size == ff_flac_blocksize_table[i]) {
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frame->blocksize = ff_flac_blocksize_table[i];
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frame->bs_code[0] = i;
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frame->bs_code[1] = 0;
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break;
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