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ffmpeg/libavcodec/alacenc.c

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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 "put_bits.h"
#include "dsputil.h"
#include "lpc.h"
#include "mathops.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
#define DEFAULT_MAX_PRED_ORDER 6
#define DEFAULT_MIN_PRED_ORDER 4
#define ALAC_MAX_LPC_PRECISION 9
#define ALAC_MAX_LPC_SHIFT 9
#define ALAC_CHMODE_LEFT_RIGHT 0
#define ALAC_CHMODE_LEFT_SIDE 1
#define ALAC_CHMODE_RIGHT_SIDE 2
#define ALAC_CHMODE_MID_SIDE 3
typedef struct RiceContext {
int history_mult;
int initial_history;
int k_modifier;
int rice_modifier;
} RiceContext;
typedef struct LPCContext {
int lpc_order;
int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
int lpc_quant;
} LPCContext;
typedef struct AlacEncodeContext {
int compression_level;
int min_prediction_order;
int max_prediction_order;
int max_coded_frame_size;
int write_sample_size;
int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
int32_t predictor_buf[DEFAULT_FRAME_SIZE];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
RiceContext rc;
LPCContext lpc[MAX_CHANNELS];
DSPContext dspctx;
AVCodecContext *avctx;
} AlacEncodeContext;
static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
{
int ch, i;
for(ch=0;ch<s->avctx->channels;ch++) {
int16_t *sptr = input_samples + ch;
for(i=0;i<s->avctx->frame_size;i++) {
s->sample_buf[ch][i] = *sptr;
sptr += s->avctx->channels;
}
}
}
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->avctx->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 calc_predictor_params(AlacEncodeContext *s, int ch)
{
int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
int shift[MAX_LPC_ORDER];
int opt_order;
opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], s->avctx->frame_size, s->min_prediction_order, s->max_prediction_order,
ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
s->lpc[ch].lpc_order = opt_order;
s->lpc[ch].lpc_quant = shift[opt_order-1];
memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
}
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];
/* calculate sum of 2nd order residual 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] += FFABS((lt + rt) >> 1);
sum[3] += FFABS(lt - rt);
sum[0] += FFABS(lt);
sum[1] += FFABS(rt);
}
/* 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;
}
}
return best;
}
static void alac_stereo_decorrelation(AlacEncodeContext *s)
{
int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
int i, mode, n = s->avctx->frame_size;
int32_t tmp;
mode = estimate_stereo_mode(left, right, n);
switch(mode)
{
case ALAC_CHMODE_LEFT_RIGHT:
s->interlacing_leftweight = 0;
s->interlacing_shift = 0;
break;
case ALAC_CHMODE_LEFT_SIDE:
for(i=0; i<n; i++) {
right[i] = left[i] - right[i];
}
s->interlacing_leftweight = 1;
s->interlacing_shift = 0;
break;
case ALAC_CHMODE_RIGHT_SIDE:
for(i=0; i<n; i++) {
tmp = right[i];
right[i] = left[i] - right[i];
left[i] = tmp + (right[i] >> 31);
}
s->interlacing_leftweight = 1;
s->interlacing_shift = 31;
break;
default:
for(i=0; i<n; i++) {
tmp = left[i];
left[i] = (tmp + right[i]) >> 1;
right[i] = tmp - right[i];
}
s->interlacing_leftweight = 1;
s->interlacing_shift = 1;
break;
}
}
static void alac_linear_predictor(AlacEncodeContext *s, int ch)
{
int i;
LPCContext lpc = s->lpc[ch];
if(lpc.lpc_order == 31) {
s->predictor_buf[0] = s->sample_buf[ch][0];
for(i=1; i<s->avctx->frame_size; i++)
s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
return;
}
// generalised linear predictor
if(lpc.lpc_order > 0) {
int32_t *samples = s->sample_buf[ch];
int32_t *residual = s->predictor_buf;
// generate warm-up samples
residual[0] = samples[0];
for(i=1;i<=lpc.lpc_order;i++)
residual[i] = samples[i] - samples[i-1];
// perform lpc on remaining samples
for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
for (j = 0; j < lpc.lpc_order; j++) {
sum += (samples[lpc.lpc_order-j] - samples[0]) *
lpc.lpc_coeff[j];
}
sum >>= lpc.lpc_quant;
sum += samples[0];
residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
s->write_sample_size);
res_val = residual[i];
if(res_val) {
int index = lpc.lpc_order - 1;
int neg = (res_val < 0);
while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
int val = samples[0] - samples[lpc.lpc_order - index];
int sign = (val ? FFSIGN(val) : 0);
if(neg)
sign*=-1;
lpc.lpc_coeff[index] -= sign;
val *= sign;
res_val -= ((val >> lpc.lpc_quant) *
(lpc.lpc_order - index));
index--;
}
}
samples++;
}
}
}
static void alac_entropy_coder(AlacEncodeContext *s)
{
unsigned int history = s->rc.initial_history;
int sign_modifier = 0, i, k;
int32_t *samples = s->predictor_buf;
for(i=0;i < s->avctx->frame_size;) {
int x;
k = av_log2((history >> 9) + 3);
x = -2*(*samples)-1;
x ^= (x>>31);
samples++;
i++;
encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
history += x * s->rc.history_mult
- ((history * s->rc.history_mult) >> 9);
sign_modifier = 0;
if(x > 0xFFFF)
history = 0xFFFF;
if((history < 128) && (i < s->avctx->frame_size)) {
unsigned int block_size = 0;
k = 7 - av_log2(history) + ((history + 16) >> 6);
while((*samples == 0) && (i < s->avctx->frame_size)) {
samples++;
i++;
block_size++;
}
encode_scalar(s, block_size, k, 16);
sign_modifier = (block_size <= 0xFFFF);
history = 0;
}
}
}
static void write_compressed_frame(AlacEncodeContext *s)
{
int i, j;
/* only simple mid/side decorrelation supported as of now */
if(s->avctx->channels == 2)
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->avctx->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->avctx->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_coded_sample = DEFAULT_SAMPLE_SIZE;
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*avctx->channels*avctx->bits_per_coded_sample)>>3;
s->write_sample_size = avctx->bits_per_coded_sample + avctx->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_coded_sample);
AV_WB8 (alac_extradata+21, avctx->channels);
AV_WB32(alac_extradata+24, s->max_coded_frame_size);
AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
AV_WB32(alac_extradata+32, avctx->sample_rate);
// 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);
}
s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
if(avctx->min_prediction_order >= 0) {
if(avctx->min_prediction_order < MIN_LPC_ORDER ||
avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
return -1;
}
s->min_prediction_order = avctx->min_prediction_order;
}
s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
if(avctx->max_prediction_order >= 0) {
if(avctx->max_prediction_order < MIN_LPC_ORDER ||
avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
return -1;
}
s->max_prediction_order = avctx->max_prediction_order;
}
if(s->max_prediction_order < s->min_prediction_order) {
av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
s->min_prediction_order, s->max_prediction_order);
return -1;
}
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);
return 0;
}
static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
{
AlacEncodeContext *s = avctx->priv_data;
PutBitContext *pb = &s->pbctx;
int i, out_bytes, verbatim_flag = 0;
if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
return -1;
}
if(buf_size < 2*s->max_coded_frame_size) {
av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
return -1;
}
verbatim:
init_put_bits(pb, frame, buf_size);
if((s->compression_level == 0) || verbatim_flag) {
// Verbatim mode
int16_t *samples = data;
write_frame_header(s, 1);
for(i=0; i<avctx->frame_size*avctx->channels; i++) {
put_sbits(pb, 16, *samples++);
}
} else {
init_sample_buffers(s, data);
write_frame_header(s, 0);
write_compressed_frame(s);
}
put_bits(pb, 3, 7);
flush_put_bits(pb);
out_bytes = put_bits_count(pb) >> 3;
if(out_bytes > s->max_coded_frame_size) {
/* frame too large. use verbatim mode */
if(verbatim_flag || (s->compression_level == 0)) {
/* still too large. must be an error. */
av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
return -1;
}
verbatim_flag = 1;
goto verbatim;
}
return out_bytes;
}
static av_cold int alac_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
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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