6c35b4dee3
Originally committed as revision 5527 to svn://svn.ffmpeg.org/ffmpeg/trunk
550 lines
13 KiB
C
550 lines
13 KiB
C
/**
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* FLAC audio encoder
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* Copyright (c) 2006 Justin Ruggles <jruggle@earthlink.net>
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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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#include "avcodec.h"
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#include "bitstream.h"
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#include "crc.h"
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#include "golomb.h"
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#define FLAC_MAX_CH 8
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#define FLAC_MIN_BLOCKSIZE 16
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#define FLAC_MAX_BLOCKSIZE 65535
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#define FLAC_SUBFRAME_CONSTANT 0
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#define FLAC_SUBFRAME_VERBATIM 1
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#define FLAC_SUBFRAME_FIXED 8
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#define FLAC_SUBFRAME_LPC 32
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#define FLAC_CHMODE_NOT_STEREO 0
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#define FLAC_CHMODE_LEFT_RIGHT 1
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#define FLAC_CHMODE_LEFT_SIDE 8
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#define FLAC_CHMODE_RIGHT_SIDE 9
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#define FLAC_CHMODE_MID_SIDE 10
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#define FLAC_STREAMINFO_SIZE 34
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typedef struct FlacSubframe {
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int type;
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int type_code;
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int obits;
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int order;
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int32_t samples[FLAC_MAX_BLOCKSIZE];
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int32_t residual[FLAC_MAX_BLOCKSIZE];
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} FlacSubframe;
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typedef struct FlacFrame {
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FlacSubframe subframes[FLAC_MAX_CH];
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int blocksize;
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int bs_code[2];
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uint8_t crc8;
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int ch_mode;
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} FlacFrame;
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typedef struct FlacEncodeContext {
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PutBitContext pb;
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int channels;
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int ch_code;
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int samplerate;
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int sr_code[2];
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int blocksize;
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int max_framesize;
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uint32_t frame_count;
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FlacFrame frame;
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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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static void write_streaminfo(FlacEncodeContext *s, uint8_t *header)
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{
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PutBitContext pb;
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memset(header, 0, FLAC_STREAMINFO_SIZE);
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init_put_bits(&pb, header, FLAC_STREAMINFO_SIZE);
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/* streaminfo metadata block */
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put_bits(&pb, 16, s->blocksize);
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put_bits(&pb, 16, s->blocksize);
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put_bits(&pb, 24, 0);
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put_bits(&pb, 24, s->max_framesize);
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put_bits(&pb, 20, s->samplerate);
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put_bits(&pb, 3, s->channels-1);
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put_bits(&pb, 5, 15); /* bits per sample - 1 */
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flush_put_bits(&pb);
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/* total samples = 0 */
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/* MD5 signature = 0 */
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}
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#define BLOCK_TIME_MS 105
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/**
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* Sets blocksize based on samplerate
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* Chooses the closest predefined blocksize >= BLOCK_TIME_MS milliseconds
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*/
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static int select_blocksize(int samplerate)
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{
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int i;
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int target;
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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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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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}
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}
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return blocksize;
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}
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static int flac_encode_init(AVCodecContext *avctx)
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{
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int freq = avctx->sample_rate;
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int channels = avctx->channels;
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FlacEncodeContext *s = avctx->priv_data;
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int i;
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uint8_t *streaminfo;
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if(avctx->sample_fmt != SAMPLE_FMT_S16) {
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return -1;
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}
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if(channels < 1 || channels > FLAC_MAX_CH) {
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return -1;
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}
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s->channels = channels;
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s->ch_code = s->channels-1;
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/* find samplerate in table */
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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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s->sr_code[0] = i;
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s->sr_code[1] = 0;
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break;
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}
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}
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/* if not in table, samplerate is non-standard */
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if(i == 12) {
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if(freq % 1000 == 0 && freq < 255000) {
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s->sr_code[0] = 12;
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s->sr_code[1] = freq / 1000;
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} else if(freq % 10 == 0 && freq < 655350) {
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s->sr_code[0] = 14;
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s->sr_code[1] = freq / 10;
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} else if(freq < 65535) {
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s->sr_code[0] = 13;
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s->sr_code[1] = freq;
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} else {
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return -1;
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}
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s->samplerate = freq;
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}
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s->blocksize = select_blocksize(s->samplerate);
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avctx->frame_size = s->blocksize;
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s->max_framesize = 14 + (s->blocksize * s->channels * 2);
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streaminfo = av_malloc(FLAC_STREAMINFO_SIZE);
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write_streaminfo(s, streaminfo);
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avctx->extradata = streaminfo;
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avctx->extradata_size = FLAC_STREAMINFO_SIZE;
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s->frame_count = 0;
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avctx->coded_frame = avcodec_alloc_frame();
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avctx->coded_frame->key_frame = 1;
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return 0;
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}
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static int init_frame(FlacEncodeContext *s)
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{
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int i, ch;
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FlacFrame *frame;
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frame = &s->frame;
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for(i=0; i<16; i++) {
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if(s->blocksize == flac_blocksizes[i]) {
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frame->blocksize = flac_blocksizes[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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}
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}
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if(i == 16) {
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frame->blocksize = s->blocksize;
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if(frame->blocksize <= 256) {
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frame->bs_code[0] = 6;
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frame->bs_code[1] = frame->blocksize-1;
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} else {
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frame->bs_code[0] = 7;
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frame->bs_code[1] = frame->blocksize-1;
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}
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}
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for(ch=0; ch<s->channels; ch++) {
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frame->subframes[ch].obits = 16;
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}
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if(s->channels == 2) {
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frame->ch_mode = FLAC_CHMODE_LEFT_RIGHT;
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} else {
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frame->ch_mode = FLAC_CHMODE_NOT_STEREO;
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}
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return 0;
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}
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/**
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* Copy channel-interleaved input samples into separate subframes
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*/
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static void copy_samples(FlacEncodeContext *s, int16_t *samples)
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{
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int i, j, ch;
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FlacFrame *frame;
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frame = &s->frame;
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for(i=0,j=0; i<frame->blocksize; i++) {
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for(ch=0; ch<s->channels; ch++,j++) {
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frame->subframes[ch].samples[i] = samples[j];
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}
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}
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}
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static void encode_residual_verbatim(FlacEncodeContext *s, int ch)
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{
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FlacFrame *frame;
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FlacSubframe *sub;
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int32_t *res;
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int32_t *smp;
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int n;
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frame = &s->frame;
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sub = &frame->subframes[ch];
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res = sub->residual;
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smp = sub->samples;
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n = frame->blocksize;
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sub->order = 0;
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sub->type = FLAC_SUBFRAME_VERBATIM;
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sub->type_code = sub->type;
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memcpy(res, smp, n * sizeof(int32_t));
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}
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static void encode_residual_fixed(int32_t *res, int32_t *smp, int n, int order)
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{
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int i;
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for(i=0; i<order; i++) {
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res[i] = smp[i];
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}
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if(order==0){
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for(i=order; i<n; i++)
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res[i]= smp[i];
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}else if(order==1){
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for(i=order; i<n; i++)
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res[i]= smp[i] - smp[i-1];
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}else if(order==2){
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for(i=order; i<n; i++)
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res[i]= smp[i] - 2*smp[i-1] + smp[i-2];
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}else if(order==3){
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for(i=order; i<n; i++)
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res[i]= smp[i] - 3*smp[i-1] + 3*smp[i-2] - smp[i-3];
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}else{
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for(i=order; i<n; i++)
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res[i]= smp[i] - 4*smp[i-1] + 6*smp[i-2] - 4*smp[i-3] + smp[i-4];
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}
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}
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static void encode_residual(FlacEncodeContext *s, int ch)
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{
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FlacFrame *frame;
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FlacSubframe *sub;
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int32_t *res;
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int32_t *smp;
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int n;
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frame = &s->frame;
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sub = &frame->subframes[ch];
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res = sub->residual;
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smp = sub->samples;
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n = frame->blocksize;
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sub->order = 2;
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sub->type = FLAC_SUBFRAME_FIXED;
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sub->type_code = sub->type | sub->order;
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encode_residual_fixed(res, smp, n, sub->order);
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}
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static void
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put_sbits(PutBitContext *pb, int bits, int32_t val)
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{
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assert(bits >= 0 && bits <= 31);
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put_bits(pb, bits, val & ((1<<bits)-1));
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}
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static void
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write_utf8(PutBitContext *pb, uint32_t val)
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{
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int bytes, shift;
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if(val < 0x80){
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put_bits(pb, 8, val);
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return;
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}
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bytes= (av_log2(val)+4) / 5;
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shift = (bytes - 1) * 6;
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put_bits(pb, 8, (256 - (256>>bytes)) | (val >> shift));
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while(shift >= 6){
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shift -= 6;
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put_bits(pb, 8, 0x80 | ((val >> shift) & 0x3F));
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}
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}
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static void
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output_frame_header(FlacEncodeContext *s)
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{
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FlacFrame *frame;
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int crc;
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frame = &s->frame;
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put_bits(&s->pb, 16, 0xFFF8);
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put_bits(&s->pb, 4, frame->bs_code[0]);
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put_bits(&s->pb, 4, s->sr_code[0]);
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if(frame->ch_mode == FLAC_CHMODE_NOT_STEREO) {
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put_bits(&s->pb, 4, s->ch_code);
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} else {
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put_bits(&s->pb, 4, frame->ch_mode);
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}
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put_bits(&s->pb, 3, 4); /* bits-per-sample code */
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put_bits(&s->pb, 1, 0);
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write_utf8(&s->pb, s->frame_count);
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if(frame->bs_code[1] > 0) {
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if(frame->bs_code[1] < 256) {
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put_bits(&s->pb, 8, frame->bs_code[1]);
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} else {
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put_bits(&s->pb, 16, frame->bs_code[1]);
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}
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}
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if(s->sr_code[1] > 0) {
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if(s->sr_code[1] < 256) {
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put_bits(&s->pb, 8, s->sr_code[1]);
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} else {
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put_bits(&s->pb, 16, s->sr_code[1]);
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}
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}
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flush_put_bits(&s->pb);
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crc = av_crc(av_crc07, 0, s->pb.buf, put_bits_count(&s->pb)>>3);
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put_bits(&s->pb, 8, crc);
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}
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static void output_subframe_verbatim(FlacEncodeContext *s, int ch)
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{
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int i;
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FlacFrame *frame;
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FlacSubframe *sub;
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int32_t res;
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frame = &s->frame;
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sub = &frame->subframes[ch];
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for(i=0; i<frame->blocksize; i++) {
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res = sub->residual[i];
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put_sbits(&s->pb, sub->obits, res);
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}
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}
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static void
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output_residual(FlacEncodeContext *ctx, int ch)
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{
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int i, j, p;
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int k, porder, psize, res_cnt;
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FlacFrame *frame;
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FlacSubframe *sub;
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frame = &ctx->frame;
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sub = &frame->subframes[ch];
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/* rice-encoded block */
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put_bits(&ctx->pb, 2, 0);
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/* partition order */
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porder = 0;
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psize = frame->blocksize;
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//porder = sub->rc.porder;
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//psize = frame->blocksize >> porder;
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put_bits(&ctx->pb, 4, porder);
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res_cnt = psize - sub->order;
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/* residual */
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j = sub->order;
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for(p=0; p<(1 << porder); p++) {
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//k = sub->rc.params[p];
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k = 9;
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put_bits(&ctx->pb, 4, k);
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if(p == 1) res_cnt = psize;
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for(i=0; i<res_cnt && j<frame->blocksize; i++, j++) {
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set_sr_golomb_flac(&ctx->pb, sub->residual[j], k, INT32_MAX, 0);
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}
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}
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}
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static void
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output_subframe_fixed(FlacEncodeContext *ctx, int ch)
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{
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int i;
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FlacFrame *frame;
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FlacSubframe *sub;
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frame = &ctx->frame;
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sub = &frame->subframes[ch];
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/* warm-up samples */
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for(i=0; i<sub->order; i++) {
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put_sbits(&ctx->pb, sub->obits, sub->residual[i]);
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}
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/* residual */
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output_residual(ctx, ch);
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}
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static void output_subframes(FlacEncodeContext *s)
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{
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FlacFrame *frame;
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FlacSubframe *sub;
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int ch;
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frame = &s->frame;
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for(ch=0; ch<s->channels; ch++) {
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sub = &frame->subframes[ch];
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/* subframe header */
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put_bits(&s->pb, 1, 0);
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put_bits(&s->pb, 6, sub->type_code);
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put_bits(&s->pb, 1, 0); /* no wasted bits */
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/* subframe */
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if(sub->type == FLAC_SUBFRAME_VERBATIM) {
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output_subframe_verbatim(s, ch);
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} else {
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output_subframe_fixed(s, ch);
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}
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}
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}
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static void output_frame_footer(FlacEncodeContext *s)
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{
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int crc;
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flush_put_bits(&s->pb);
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crc = bswap_16(av_crc(av_crc8005, 0, s->pb.buf, put_bits_count(&s->pb)>>3));
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put_bits(&s->pb, 16, crc);
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flush_put_bits(&s->pb);
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}
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static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
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int buf_size, void *data)
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{
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int ch;
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FlacEncodeContext *s;
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int16_t *samples = data;
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int out_bytes;
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s = avctx->priv_data;
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s->blocksize = avctx->frame_size;
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if(init_frame(s)) {
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return 0;
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}
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copy_samples(s, samples);
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for(ch=0; ch<s->channels; ch++) {
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encode_residual(s, ch);
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}
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init_put_bits(&s->pb, frame, buf_size);
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output_frame_header(s);
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output_subframes(s);
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output_frame_footer(s);
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out_bytes = put_bits_count(&s->pb) >> 3;
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if(out_bytes > s->max_framesize || out_bytes >= buf_size) {
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/* frame too large. use verbatim mode */
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for(ch=0; ch<s->channels; ch++) {
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encode_residual_verbatim(s, ch);
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}
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init_put_bits(&s->pb, frame, buf_size);
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output_frame_header(s);
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output_subframes(s);
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output_frame_footer(s);
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out_bytes = put_bits_count(&s->pb) >> 3;
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if(out_bytes > s->max_framesize || out_bytes >= buf_size) {
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/* still too large. must be an error. */
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av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
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return -1;
|
|
}
|
|
}
|
|
|
|
s->frame_count++;
|
|
return out_bytes;
|
|
}
|
|
|
|
static int flac_encode_close(AVCodecContext *avctx)
|
|
{
|
|
av_freep(&avctx->coded_frame);
|
|
return 0;
|
|
}
|
|
|
|
AVCodec flac_encoder = {
|
|
"flac",
|
|
CODEC_TYPE_AUDIO,
|
|
CODEC_ID_FLAC,
|
|
sizeof(FlacEncodeContext),
|
|
flac_encode_init,
|
|
flac_encode_frame,
|
|
flac_encode_close,
|
|
NULL,
|
|
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
|
|
};
|