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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 "put_bits.h"
#include "dsputil.h"
#include "internal.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 AlacLPCContext {
int lpc_order;
int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
int lpc_quant;
} AlacLPCContext;
typedef struct AlacEncodeContext {
int frame_size; /**< current frame size */
int verbatim; /**< current frame verbatim mode flag */
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;
AlacLPCContext lpc[MAX_CHANNELS];
LPCContext lpc_ctx;
AVCodecContext *avctx;
} AlacEncodeContext;
static void init_sample_buffers(AlacEncodeContext *s,
const int16_t *input_samples)
{
int ch, i;
for (ch = 0; ch < s->avctx->channels; ch++) {
const int16_t *sptr = input_samples + ch;
for (i = 0; i < s->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 encode_fs = 0;
if (s->frame_size < DEFAULT_FRAME_SIZE)
encode_fs = 1;
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, encode_fs); // Sample count is in the header
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
if (encode_fs)
put_bits32(&s->pbctx, s->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;
if (s->compression_level == 1) {
s->lpc[ch].lpc_order = 6;
s->lpc[ch].lpc_quant = 6;
s->lpc[ch].lpc_coeff[0] = 160;
s->lpc[ch].lpc_coeff[1] = -190;
s->lpc[ch].lpc_coeff[2] = 170;
s->lpc[ch].lpc_coeff[3] = -130;
s->lpc[ch].lpc_coeff[4] = 80;
s->lpc[ch].lpc_coeff[5] = -25;
} else {
opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
s->frame_size,
s->min_prediction_order,
s->max_prediction_order,
ALAC_MAX_LPC_PRECISION, coefs, shift,
FF_LPC_TYPE_LEVINSON, 0,
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->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;
AlacLPCContext lpc = s->lpc[ch];
if (lpc.lpc_order == 31) {
s->predictor_buf[0] = s->sample_buf[ch][0];
for (i = 1; i < s->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->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->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->frame_size) {
unsigned int block_size = 0;
k = 7 - av_log2(history) + ((history + 16) >> 6);
while (*samples == 0 && i < s->frame_size) {
samples++;
i++;
block_size++;
}
encode_scalar(s, block_size, k, 16);
sign_modifier = (block_size <= 0xFFFF);
history = 0;
}
}
}
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
const int16_t *samples)
{
int i, j;
int prediction_type = 0;
PutBitContext *pb = &s->pbctx;
init_put_bits(pb, avpkt->data, avpkt->size);
if (s->verbatim) {
write_frame_header(s);
for (i = 0; i < s->frame_size * s->avctx->channels; i++)
put_sbits(pb, 16, *samples++);
} else {
init_sample_buffers(s, samples);
write_frame_header(s);
if (s->avctx->channels == 2)
alac_stereo_decorrelation(s);
put_bits(pb, 8, s->interlacing_shift);
put_bits(pb, 8, s->interlacing_leftweight);
for (i = 0; i < s->avctx->channels; i++) {
calc_predictor_params(s, i);
put_bits(pb, 4, prediction_type);
put_bits(pb, 4, s->lpc[i].lpc_quant);
put_bits(pb, 3, s->rc.rice_modifier);
put_bits(pb, 5, s->lpc[i].lpc_order);
// predictor coeff. table
for (j = 0; j < s->lpc[i].lpc_order; j++)
put_sbits(pb, 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);
// TODO: determine when this will actually help. for now it's not used.
if (prediction_type == 15) {
// 2nd pass 1st order filter
for (j = s->frame_size - 1; j > 0; j--)
s->predictor_buf[j] -= s->predictor_buf[j - 1];
}
alac_entropy_coder(s);
}
}
put_bits(pb, 3, 7);
flush_put_bits(pb);
return put_bits_count(pb) >> 3;
}
static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
{
int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE);
return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8;
}
static av_cold int alac_encode_close(AVCodecContext *avctx)
{
AlacEncodeContext *s = avctx->priv_data;
ff_lpc_end(&s->lpc_ctx);
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
return 0;
}
static av_cold int alac_encode_init(AVCodecContext *avctx)
{
AlacEncodeContext *s = avctx->priv_data;
int ret;
uint8_t *alac_extradata;
avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
return -1;
}
/* TODO: Correctly implement multi-channel ALAC.
It is similar to multi-channel AAC, in that it has a series of
single-channel (SCE), channel-pair (CPE), and LFE elements. */
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "only mono or stereo input is currently supported\n");
return AVERROR_PATCHWELCOME;
}
// Set default compression level
if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
s->compression_level = 2;
else
s->compression_level = av_clip(avctx->compression_level, 0, 2);
// 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 = get_max_frame_size(avctx->frame_size,
avctx->channels,
DEFAULT_SAMPLE_SIZE);
// FIXME: consider wasted_bytes
s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1;
avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
ret = AVERROR(ENOMEM);
goto error;
}
avctx->extradata_size = ALAC_EXTRADATA_SIZE;
alac_extradata = avctx->extradata;
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, DEFAULT_SAMPLE_SIZE);
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 * DEFAULT_SAMPLE_SIZE); // 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);
ret = AVERROR(EINVAL);
goto error;
}
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);
ret = AVERROR(EINVAL);
goto error;
}
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);
ret = AVERROR(EINVAL);
goto error;
}
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
s->avctx = avctx;
if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
s->max_prediction_order,
FF_LPC_TYPE_LEVINSON)) < 0) {
goto error;
}
return 0;
error:
alac_encode_close(avctx);
return ret;
}
static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AlacEncodeContext *s = avctx->priv_data;
int out_bytes, max_frame_size, ret;
const int16_t *samples = (const int16_t *)frame->data[0];
s->frame_size = frame->nb_samples;
if (avctx->frame_size < DEFAULT_FRAME_SIZE)
max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
DEFAULT_SAMPLE_SIZE);
else
max_frame_size = s->max_coded_frame_size;
if ((ret = ff_alloc_packet(avpkt, 2 * max_frame_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
/* use verbatim mode for compression_level 0 */
s->verbatim = !s->compression_level;
out_bytes = write_frame(s, avpkt, samples);
if (out_bytes > max_frame_size) {
/* frame too large. use verbatim mode */
s->verbatim = 1;
out_bytes = write_frame(s, avpkt, samples);
}
avpkt->size = out_bytes;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_alac_encoder = {
.name = "alac",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ALAC,
.priv_data_size = sizeof(AlacEncodeContext),
.init = alac_encode_init,
.encode2 = alac_encode_frame,
.close = alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
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