mirror of
https://github.com/xenia-project/FFmpeg.git
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e71bcc3798
1) search for optimal rice parameters and partition order. i also modified the stereo method estimation to use this to calculate estimated bit count instead of using just the pure sums. 2) search for the best fixed prediction order 3) constant subframe mode (good for encoding silence) Note that the regression test for the decoded wav file also changed. This is due to FFmpeg's FLAC decoder truncating the file, which it did before anyway...just at a different cutoff point. The generated FLAC files are still 100% lossless. With this update, FFmpeg's FLAC encoder has speed and compression somewhere between "flac -1" and "flac -2". On my machine, it's about 15% faster than "flac -2", and about 10% slower than "flac -1". The encoding parameters are identical to "flac -2" (fixed predictors, 1152 blocksize, partition order 0 to 3). Originally committed as revision 5536 to svn://svn.ffmpeg.org/ffmpeg/trunk
838 lines
20 KiB
C
838 lines
20 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 RiceContext {
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int porder;
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int params[256];
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} RiceContext;
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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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RiceContext rc;
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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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AVCodecContext *avctx;
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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 27
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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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s->avctx = avctx;
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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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/* set maximum encoded frame size in verbatim mode */
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if(s->channels == 2) {
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s->max_framesize = 14 + ((s->blocksize * 33 + 7) >> 3);
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} else {
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s->max_framesize = 14 + (s->blocksize * s->channels * 2);
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}
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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 void 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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}
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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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#define rice_encode_count(sum, n, k) (((n)*((k)+1))+((sum-(n>>1))>>(k)))
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static int find_optimal_param(uint32_t sum, int n)
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{
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int k, k_opt;
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uint32_t nbits, nbits_opt;
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k_opt = 0;
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nbits_opt = rice_encode_count(sum, n, 0);
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for(k=1; k<=14; k++) {
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nbits = rice_encode_count(sum, n, k);
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if(nbits < nbits_opt) {
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nbits_opt = nbits;
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k_opt = k;
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}
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}
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return k_opt;
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}
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static uint32_t calc_optimal_rice_params(RiceContext *rc, int porder,
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uint32_t *sums, int n, int pred_order)
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{
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int i;
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int k, cnt, part;
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uint32_t all_bits;
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part = (1 << porder);
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all_bits = 0;
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cnt = (n >> porder) - pred_order;
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for(i=0; i<part; i++) {
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if(i == 1) cnt = (n >> porder);
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k = find_optimal_param(sums[i], cnt);
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rc->params[i] = k;
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all_bits += rice_encode_count(sums[i], cnt, k);
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}
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all_bits += (4 * part);
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rc->porder = porder;
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return all_bits;
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}
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static void calc_sums(int pmax, uint32_t *data, int n, int pred_order,
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uint32_t sums[][256])
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{
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int i, j;
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int parts, cnt;
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uint32_t *res;
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/* sums for highest level */
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parts = (1 << pmax);
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res = &data[pred_order];
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cnt = (n >> pmax) - pred_order;
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for(i=0; i<parts; i++) {
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if(i == 1) cnt = (n >> pmax);
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if(i > 0) res = &data[i*cnt];
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sums[pmax][i] = 0;
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for(j=0; j<cnt; j++) {
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sums[pmax][i] += res[j];
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}
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}
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/* sums for lower levels */
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for(i=pmax-1; i>=0; i--) {
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parts = (1 << i);
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for(j=0; j<parts; j++) {
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sums[i][j] = sums[i+1][2*j] + sums[i+1][2*j+1];
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}
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}
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}
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static uint32_t calc_rice_params(RiceContext *rc, int pmax, int32_t *data,
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int n, int pred_order)
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{
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int i;
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uint32_t bits, opt_bits;
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int opt_porder;
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RiceContext opt_rc;
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uint32_t *udata;
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uint32_t sums[9][256];
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assert(pmax >= 0 && pmax <= 8);
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udata = av_malloc(n * sizeof(uint32_t));
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for(i=0; i<n; i++) {
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udata[i] = (2*data[i]) ^ (data[i]>>31);
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}
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calc_sums(pmax, udata, n, pred_order, sums);
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opt_porder = 0;
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opt_bits = UINT32_MAX;
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for(i=0; i<=pmax; i++) {
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bits = calc_optimal_rice_params(rc, i, sums[i], n, pred_order);
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if(bits < opt_bits) {
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opt_bits = bits;
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opt_porder = i;
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memcpy(&opt_rc, rc, sizeof(RiceContext));
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}
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}
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if(opt_porder != pmax) {
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memcpy(rc, &opt_rc, sizeof(RiceContext));
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}
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av_freep(&udata);
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return opt_bits;
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}
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static uint32_t calc_rice_params_fixed(RiceContext *rc, int pmax, int32_t *data,
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int n, int pred_order, int bps)
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{
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uint32_t bits;
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bits = pred_order*bps + 6;
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bits += calc_rice_params(rc, pmax, data, n, pred_order);
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return bits;
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}
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static void encode_residual_verbatim(int32_t *res, int32_t *smp, int n)
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{
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assert(n > 0);
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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 int get_max_p_order(int max_porder, int n, int order)
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{
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int porder, max_parts;
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porder = max_porder;
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while(porder > 0) {
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max_parts = (1 << porder);
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if(!(n % max_parts) && (n > max_parts*order)) {
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break;
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}
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porder--;
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}
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return porder;
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}
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static int encode_residual(FlacEncodeContext *ctx, int ch)
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{
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int i, opt_order, porder, max_porder, n;
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FlacFrame *frame;
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FlacSubframe *sub;
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uint32_t bits[5];
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int32_t *res, *smp;
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frame = &ctx->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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/* CONSTANT */
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for(i=1; i<n; i++) {
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if(smp[i] != smp[0]) break;
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}
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if(i == n) {
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sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT;
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res[0] = smp[0];
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return sub->obits;
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}
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/* VERBATIM */
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if(n < 5) {
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sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM;
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encode_residual_verbatim(res, smp, n);
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return sub->obits * n;
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}
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max_porder = 3;
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/* FIXED */
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opt_order = 0;
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bits[0] = UINT32_MAX;
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for(i=0; i<=4; i++) {
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encode_residual_fixed(res, smp, n, i);
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porder = get_max_p_order(max_porder, n, i);
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bits[i] = calc_rice_params_fixed(&sub->rc, porder, res, n, i, sub->obits);
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if(bits[i] < bits[opt_order]) {
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opt_order = i;
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}
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}
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sub->order = opt_order;
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sub->type = FLAC_SUBFRAME_FIXED;
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sub->type_code = sub->type | sub->order;
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if(sub->order != 4) {
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encode_residual_fixed(res, smp, n, sub->order);
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porder = get_max_p_order(max_porder, n, sub->order);
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calc_rice_params_fixed(&sub->rc, porder, res, n, sub->order, sub->obits);
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}
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return bits[sub->order];
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}
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static int encode_residual_v(FlacEncodeContext *ctx, int ch)
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{
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int i, n;
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FlacFrame *frame;
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FlacSubframe *sub;
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int32_t *res, *smp;
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frame = &ctx->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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/* CONSTANT */
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for(i=1; i<n; i++) {
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if(smp[i] != smp[0]) break;
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}
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if(i == n) {
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sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT;
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res[0] = smp[0];
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return sub->obits;
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}
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/* VERBATIM */
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sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM;
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encode_residual_verbatim(res, smp, n);
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return sub->obits * n;
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}
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|
|
static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
|
|
{
|
|
int i, best;
|
|
int32_t lt, rt;
|
|
uint64_t sum[4];
|
|
uint64_t score[4];
|
|
int k;
|
|
|
|
/* calculate sum of squares for each channel */
|
|
sum[0] = sum[1] = sum[2] = sum[3] = 0;
|
|
for(i=2; i<n; i++) {
|
|
lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
|
|
rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
|
|
sum[2] += ABS((lt + rt) >> 1);
|
|
sum[3] += ABS(lt - rt);
|
|
sum[0] += ABS(lt);
|
|
sum[1] += ABS(rt);
|
|
}
|
|
for(i=0; i<4; i++) {
|
|
k = find_optimal_param(2*sum[i], n);
|
|
sum[i] = rice_encode_count(2*sum[i], n, k);
|
|
}
|
|
|
|
/* calculate score for each mode */
|
|
score[0] = sum[0] + sum[1];
|
|
score[1] = sum[0] + sum[3];
|
|
score[2] = sum[1] + sum[3];
|
|
score[3] = sum[2] + sum[3];
|
|
|
|
/* return mode with lowest score */
|
|
best = 0;
|
|
for(i=1; i<4; i++) {
|
|
if(score[i] < score[best]) {
|
|
best = i;
|
|
}
|
|
}
|
|
if(best == 0) {
|
|
return FLAC_CHMODE_LEFT_RIGHT;
|
|
} else if(best == 1) {
|
|
return FLAC_CHMODE_LEFT_SIDE;
|
|
} else if(best == 2) {
|
|
return FLAC_CHMODE_RIGHT_SIDE;
|
|
} else {
|
|
return FLAC_CHMODE_MID_SIDE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Perform stereo channel decorrelation
|
|
*/
|
|
static void channel_decorrelation(FlacEncodeContext *ctx)
|
|
{
|
|
FlacFrame *frame;
|
|
int32_t *left, *right;
|
|
int i, n;
|
|
|
|
frame = &ctx->frame;
|
|
n = frame->blocksize;
|
|
left = frame->subframes[0].samples;
|
|
right = frame->subframes[1].samples;
|
|
|
|
if(ctx->channels != 2) {
|
|
frame->ch_mode = FLAC_CHMODE_NOT_STEREO;
|
|
return;
|
|
}
|
|
|
|
frame->ch_mode = estimate_stereo_mode(left, right, n);
|
|
|
|
/* perform decorrelation and adjust bits-per-sample */
|
|
if(frame->ch_mode == FLAC_CHMODE_LEFT_RIGHT) {
|
|
return;
|
|
}
|
|
if(frame->ch_mode == FLAC_CHMODE_MID_SIDE) {
|
|
int32_t tmp;
|
|
for(i=0; i<n; i++) {
|
|
tmp = left[i];
|
|
left[i] = (tmp + right[i]) >> 1;
|
|
right[i] = tmp - right[i];
|
|
}
|
|
frame->subframes[1].obits++;
|
|
} else if(frame->ch_mode == FLAC_CHMODE_LEFT_SIDE) {
|
|
for(i=0; i<n; i++) {
|
|
right[i] = left[i] - right[i];
|
|
}
|
|
frame->subframes[1].obits++;
|
|
} else {
|
|
for(i=0; i<n; i++) {
|
|
left[i] -= right[i];
|
|
}
|
|
frame->subframes[0].obits++;
|
|
}
|
|
}
|
|
|
|
static void put_sbits(PutBitContext *pb, int bits, int32_t val)
|
|
{
|
|
assert(bits >= 0 && bits <= 31);
|
|
|
|
put_bits(pb, bits, val & ((1<<bits)-1));
|
|
}
|
|
|
|
static void write_utf8(PutBitContext *pb, uint32_t val)
|
|
{
|
|
int bytes, shift;
|
|
|
|
if(val < 0x80){
|
|
put_bits(pb, 8, val);
|
|
return;
|
|
}
|
|
|
|
bytes= (av_log2(val)+4) / 5;
|
|
shift = (bytes - 1) * 6;
|
|
put_bits(pb, 8, (256 - (256>>bytes)) | (val >> shift));
|
|
while(shift >= 6){
|
|
shift -= 6;
|
|
put_bits(pb, 8, 0x80 | ((val >> shift) & 0x3F));
|
|
}
|
|
}
|
|
|
|
static void output_frame_header(FlacEncodeContext *s)
|
|
{
|
|
FlacFrame *frame;
|
|
int crc;
|
|
|
|
frame = &s->frame;
|
|
|
|
put_bits(&s->pb, 16, 0xFFF8);
|
|
put_bits(&s->pb, 4, frame->bs_code[0]);
|
|
put_bits(&s->pb, 4, s->sr_code[0]);
|
|
if(frame->ch_mode == FLAC_CHMODE_NOT_STEREO) {
|
|
put_bits(&s->pb, 4, s->ch_code);
|
|
} else {
|
|
put_bits(&s->pb, 4, frame->ch_mode);
|
|
}
|
|
put_bits(&s->pb, 3, 4); /* bits-per-sample code */
|
|
put_bits(&s->pb, 1, 0);
|
|
write_utf8(&s->pb, s->frame_count);
|
|
if(frame->bs_code[0] == 6) {
|
|
put_bits(&s->pb, 8, frame->bs_code[1]);
|
|
} else if(frame->bs_code[0] == 7) {
|
|
put_bits(&s->pb, 16, frame->bs_code[1]);
|
|
}
|
|
if(s->sr_code[0] == 12) {
|
|
put_bits(&s->pb, 8, s->sr_code[1]);
|
|
} else if(s->sr_code[0] > 12) {
|
|
put_bits(&s->pb, 16, s->sr_code[1]);
|
|
}
|
|
flush_put_bits(&s->pb);
|
|
crc = av_crc(av_crc07, 0, s->pb.buf, put_bits_count(&s->pb)>>3);
|
|
put_bits(&s->pb, 8, crc);
|
|
}
|
|
|
|
static void output_subframe_constant(FlacEncodeContext *s, int ch)
|
|
{
|
|
FlacSubframe *sub;
|
|
int32_t res;
|
|
|
|
sub = &s->frame.subframes[ch];
|
|
res = sub->residual[0];
|
|
put_sbits(&s->pb, sub->obits, res);
|
|
}
|
|
|
|
static void output_subframe_verbatim(FlacEncodeContext *s, int ch)
|
|
{
|
|
int i;
|
|
FlacFrame *frame;
|
|
FlacSubframe *sub;
|
|
int32_t res;
|
|
|
|
frame = &s->frame;
|
|
sub = &frame->subframes[ch];
|
|
|
|
for(i=0; i<frame->blocksize; i++) {
|
|
res = sub->residual[i];
|
|
put_sbits(&s->pb, sub->obits, res);
|
|
}
|
|
}
|
|
|
|
static void output_residual(FlacEncodeContext *ctx, int ch)
|
|
{
|
|
int i, j, p, n, parts;
|
|
int k, porder, psize, res_cnt;
|
|
FlacFrame *frame;
|
|
FlacSubframe *sub;
|
|
int32_t *res;
|
|
|
|
frame = &ctx->frame;
|
|
sub = &frame->subframes[ch];
|
|
res = sub->residual;
|
|
n = frame->blocksize;
|
|
|
|
/* rice-encoded block */
|
|
put_bits(&ctx->pb, 2, 0);
|
|
|
|
/* partition order */
|
|
porder = sub->rc.porder;
|
|
psize = n >> porder;
|
|
parts = (1 << porder);
|
|
put_bits(&ctx->pb, 4, porder);
|
|
res_cnt = psize - sub->order;
|
|
|
|
/* residual */
|
|
j = sub->order;
|
|
for(p=0; p<parts; p++) {
|
|
k = sub->rc.params[p];
|
|
put_bits(&ctx->pb, 4, k);
|
|
if(p == 1) res_cnt = psize;
|
|
for(i=0; i<res_cnt && j<n; i++, j++) {
|
|
set_sr_golomb_flac(&ctx->pb, res[j], k, INT32_MAX, 0);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void output_subframe_fixed(FlacEncodeContext *ctx, int ch)
|
|
{
|
|
int i;
|
|
FlacFrame *frame;
|
|
FlacSubframe *sub;
|
|
|
|
frame = &ctx->frame;
|
|
sub = &frame->subframes[ch];
|
|
|
|
/* warm-up samples */
|
|
for(i=0; i<sub->order; i++) {
|
|
put_sbits(&ctx->pb, sub->obits, sub->residual[i]);
|
|
}
|
|
|
|
/* residual */
|
|
output_residual(ctx, ch);
|
|
}
|
|
|
|
static void output_subframes(FlacEncodeContext *s)
|
|
{
|
|
FlacFrame *frame;
|
|
FlacSubframe *sub;
|
|
int ch;
|
|
|
|
frame = &s->frame;
|
|
|
|
for(ch=0; ch<s->channels; ch++) {
|
|
sub = &frame->subframes[ch];
|
|
|
|
/* subframe header */
|
|
put_bits(&s->pb, 1, 0);
|
|
put_bits(&s->pb, 6, sub->type_code);
|
|
put_bits(&s->pb, 1, 0); /* no wasted bits */
|
|
|
|
/* subframe */
|
|
if(sub->type == FLAC_SUBFRAME_CONSTANT) {
|
|
output_subframe_constant(s, ch);
|
|
} else if(sub->type == FLAC_SUBFRAME_VERBATIM) {
|
|
output_subframe_verbatim(s, ch);
|
|
} else if(sub->type == FLAC_SUBFRAME_FIXED) {
|
|
output_subframe_fixed(s, ch);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void output_frame_footer(FlacEncodeContext *s)
|
|
{
|
|
int crc;
|
|
flush_put_bits(&s->pb);
|
|
crc = bswap_16(av_crc(av_crc8005, 0, s->pb.buf, put_bits_count(&s->pb)>>3));
|
|
put_bits(&s->pb, 16, crc);
|
|
flush_put_bits(&s->pb);
|
|
}
|
|
|
|
static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
|
|
int buf_size, void *data)
|
|
{
|
|
int ch;
|
|
FlacEncodeContext *s;
|
|
int16_t *samples = data;
|
|
int out_bytes;
|
|
|
|
s = avctx->priv_data;
|
|
|
|
s->blocksize = avctx->frame_size;
|
|
init_frame(s);
|
|
|
|
copy_samples(s, samples);
|
|
|
|
channel_decorrelation(s);
|
|
|
|
for(ch=0; ch<s->channels; ch++) {
|
|
encode_residual(s, ch);
|
|
}
|
|
init_put_bits(&s->pb, frame, buf_size);
|
|
output_frame_header(s);
|
|
output_subframes(s);
|
|
output_frame_footer(s);
|
|
out_bytes = put_bits_count(&s->pb) >> 3;
|
|
|
|
if(out_bytes > s->max_framesize || out_bytes >= buf_size) {
|
|
/* frame too large. use verbatim mode */
|
|
for(ch=0; ch<s->channels; ch++) {
|
|
encode_residual_v(s, ch);
|
|
}
|
|
init_put_bits(&s->pb, frame, buf_size);
|
|
output_frame_header(s);
|
|
output_subframes(s);
|
|
output_frame_footer(s);
|
|
out_bytes = put_bits_count(&s->pb) >> 3;
|
|
|
|
if(out_bytes > s->max_framesize || out_bytes >= buf_size) {
|
|
/* still too large. must be an error. */
|
|
av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
s->frame_count++;
|
|
return out_bytes;
|
|
}
|
|
|
|
static int flac_encode_close(AVCodecContext *avctx)
|
|
{
|
|
av_freep(&avctx->extradata);
|
|
avctx->extradata_size = 0;
|
|
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,
|
|
};
|