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/**
* ALAC audio encoder
* Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "lpc.h"
#define DEFAULT_FRAME_SIZE 4096
#define DEFAULT_SAMPLE_SIZE 16
#define MAX_CHANNELS 8
#define ALAC_EXTRADATA_SIZE 36
#define ALAC_FRAME_HEADER_SIZE 55
#define ALAC_FRAME_FOOTER_SIZE 3
#define ALAC_ESCAPE_CODE 0x1FF
#define ALAC_MAX_LPC_ORDER 30
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
DSPContext dspctx;
AVCodecContext *avctx;
} AlacEncodeContext;
static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
{
int divisor, q, r;
k = FFMIN(k, s->rc.k_modifier);
divisor = (1<<k) - 1;
q = x / divisor;
r = x % divisor;
if(q > 8) {
// write escape code and sample value directly
put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
put_bits(&s->pbctx, write_sample_size, x);
} else {
if(q)
put_bits(&s->pbctx, q, (1<<q) - 1);
put_bits(&s->pbctx, 1, 0);
if(k != 1) {
if(r > 0)
put_bits(&s->pbctx, k, r+1);
else
put_bits(&s->pbctx, k-1, 0);
}
}
}
static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
{
put_bits(&s->pbctx, 3, s->channels-1); // No. of channels -1
put_bits(&s->pbctx, 16, 0); // Seems to be zero
put_bits(&s->pbctx, 1, 1); // Sample count is in the header
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
}
static void write_compressed_frame(AlacEncodeContext *s)
{
int i, j;
/* only simple mid/side decorrelation supported as of now */
alac_stereo_decorrelation(s);
put_bits(&s->pbctx, 8, s->interlacing_shift);
put_bits(&s->pbctx, 8, s->interlacing_leftweight);
for(i=0;i<s->channels;i++) {
calc_predictor_params(s, i);
put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
put_bits(&s->pbctx, 3, s->rc.rice_modifier);
put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
// predictor coeff. table
for(j=0;j<s->lpc[i].lpc_order;j++) {
put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
}
}
// apply lpc and entropy coding to audio samples
for(i=0;i<s->channels;i++) {
alac_linear_predictor(s, i);
alac_entropy_coder(s);
}
}
static av_cold int alac_encode_init(AVCodecContext *avctx)
{
AlacEncodeContext *s = avctx->priv_data;
uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
avctx->frame_size = DEFAULT_FRAME_SIZE;
avctx->bits_per_sample = DEFAULT_SAMPLE_SIZE;
s->channels = avctx->channels;
s->samplerate = avctx->sample_rate;
if(avctx->sample_fmt != SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
return -1;
}
// Set default compression level
if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
s->compression_level = 1;
else
s->compression_level = av_clip(avctx->compression_level, 0, 1);
// Initialize default Rice parameters
s->rc.history_mult = 40;
s->rc.initial_history = 10;
s->rc.k_modifier = 14;
s->rc.rice_modifier = 4;
s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
avctx->frame_size*s->channels*avctx->bits_per_sample)>>3;
s->write_sample_size = avctx->bits_per_sample + s->channels - 1; // FIXME: consider wasted_bytes
AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
AV_WB32(alac_extradata+12, avctx->frame_size);
AV_WB8 (alac_extradata+17, avctx->bits_per_sample);
AV_WB8 (alac_extradata+21, s->channels);
AV_WB32(alac_extradata+24, s->max_coded_frame_size);
AV_WB32(alac_extradata+28, s->samplerate*s->channels*avctx->bits_per_sample); // average bitrate
AV_WB32(alac_extradata+32, s->samplerate);
// Set relevant extradata fields
if(s->compression_level > 0) {
AV_WB8(alac_extradata+18, s->rc.history_mult);
AV_WB8(alac_extradata+19, s->rc.initial_history);
AV_WB8(alac_extradata+20, s->rc.k_modifier);
}
avctx->extradata = alac_extradata;
avctx->extradata_size = ALAC_EXTRADATA_SIZE;
avctx->coded_frame = avcodec_alloc_frame();
avctx->coded_frame->key_frame = 1;
s->avctx = avctx;
dsputil_init(&s->dspctx, avctx);
allocate_sample_buffers(s);
return 0;
}
static av_cold int alac_encode_close(AVCodecContext *avctx)
{
AlacEncodeContext *s = avctx->priv_data;
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
free_sample_buffers(s);
return 0;
}
AVCodec alac_encoder = {
"alac",
CODEC_TYPE_AUDIO,
CODEC_ID_ALAC,
sizeof(AlacEncodeContext),
alac_encode_init,
alac_encode_frame,
alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
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