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ffmpeg/libavcodec/alacenc.c
Andreas Rheinhardt 56e9e0273a avcodec/encode: Always use intermediate buffer in ff_alloc_packet2() 
Up until now, ff_alloc_packet2() has a min_size parameter:
It is supposed to be a lower bound on the final size of the packet
to allocate. If it is not too far from the upper bound (namely,
if it is at least half the upper bound), then ff_alloc_packet2()
already allocates the final, already refcounted packet; if it is
not, then the packet is not refcounted and its data only points to
a buffer owned by the AVCodecContext (in this case, the packet will
be made refcounted in encode_simple_internal() in libavcodec/encode.c).
The goal of this was to avoid data copies and intermediate buffers
if one has a precise lower bound.

Yet those encoders for which precise lower bounds exist have recently
been switched to ff_get_encode_buffer() (which automatically allocates
final buffers), leaving only two encoders to actually set the min_size
to something else than zero (namely aliaspixenc and hapenc). Both of
these encoders use a very low lower bound that is not helpful in any
nontrivial case.

This commit therefore removes the min_size parameter as well as the
codepath in ff_alloc_packet2() for the allocation of final buffers.
Furthermore, the function has been renamed to ff_alloc_packet() and
moved to encode.h alongside ff_get_encode_buffer().

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-06-08 12:52:50 +02:00

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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 "libavutil/opt.h"
#include "avcodec.h"
#include "encode.h"
#include "put_bits.h"
#include "internal.h"
#include "lpc.h"
#include "mathops.h"
#include "alac_data.h"
#define DEFAULT_FRAME_SIZE 4096
#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_MIN_LPC_SHIFT 0
#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 {
const AVClass *class;
AVCodecContext *avctx;
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;
int extra_bits;
int32_t sample_buf[2][DEFAULT_FRAME_SIZE];
int32_t predictor_buf[2][DEFAULT_FRAME_SIZE];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
RiceContext rc;
AlacLPCContext lpc[2];
LPCContext lpc_ctx;
} AlacEncodeContext;
static void init_sample_buffers(AlacEncodeContext *s, int channels,
const uint8_t *samples[2])
{
int ch, i;
int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
s->avctx->bits_per_raw_sample;
#define COPY_SAMPLES(type) do { \
for (ch = 0; ch < channels; ch++) { \
int32_t *bptr = s->sample_buf[ch]; \
const type *sptr = (const type *)samples[ch]; \
for (i = 0; i < s->frame_size; i++) \
bptr[i] = sptr[i] >> shift; \
} \
} while (0)
if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P)
COPY_SAMPLES(int32_t);
else
COPY_SAMPLES(int16_t);
}
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_element_header(AlacEncodeContext *s,
enum AlacRawDataBlockType element,
int instance)
{
int encode_fs = 0;
if (s->frame_size < DEFAULT_FRAME_SIZE)
encode_fs = 1;
put_bits(&s->pbctx, 3, element); // element type
put_bits(&s->pbctx, 4, instance); // element instance
put_bits(&s->pbctx, 12, 0); // unused header bits
put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)
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_MIN_LPC_SHIFT,
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];
int32_t *residual = s->predictor_buf[ch];
if (lpc.lpc_order == 31) {
residual[0] = s->sample_buf[ch][0];
for (i = 1; i < s->frame_size; i++) {
residual[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];
// generate warm-up samples
residual[0] = samples[0];
for (i = 1; i <= lpc.lpc_order; i++)
residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);
// 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, int ch)
{
unsigned int history = s->rc.initial_history;
int sign_modifier = 0, i, k;
int32_t *samples = s->predictor_buf[ch];
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 void write_element(AlacEncodeContext *s,
enum AlacRawDataBlockType element, int instance,
const uint8_t *samples0, const uint8_t *samples1)
{
const uint8_t *samples[2] = { samples0, samples1 };
int i, j, channels;
int prediction_type = 0;
PutBitContext *pb = &s->pbctx;
channels = element == TYPE_CPE ? 2 : 1;
if (s->verbatim) {
write_element_header(s, element, instance);
/* samples are channel-interleaved in verbatim mode */
if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
int shift = 32 - s->avctx->bits_per_raw_sample;
const int32_t *samples_s32[2] = { (const int32_t *)samples0,
(const int32_t *)samples1 };
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < channels; j++)
put_sbits(pb, s->avctx->bits_per_raw_sample,
samples_s32[j][i] >> shift);
} else {
const int16_t *samples_s16[2] = { (const int16_t *)samples0,
(const int16_t *)samples1 };
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < channels; j++)
put_sbits(pb, s->avctx->bits_per_raw_sample,
samples_s16[j][i]);
}
} else {
s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +
channels - 1;
init_sample_buffers(s, channels, samples);
write_element_header(s, element, instance);
// extract extra bits if needed
if (s->extra_bits) {
uint32_t mask = (1 << s->extra_bits) - 1;
for (j = 0; j < channels; j++) {
int32_t *extra = s->predictor_buf[j];
int32_t *smp = s->sample_buf[j];
for (i = 0; i < s->frame_size; i++) {
extra[i] = smp[i] & mask;
smp[i] >>= s->extra_bits;
}
}
}
if (channels == 2)
alac_stereo_decorrelation(s);
else
s->interlacing_shift = s->interlacing_leftweight = 0;
put_bits(pb, 8, s->interlacing_shift);
put_bits(pb, 8, s->interlacing_leftweight);
for (i = 0; i < 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]);
}
// write extra bits if needed
if (s->extra_bits) {
for (i = 0; i < s->frame_size; i++) {
for (j = 0; j < channels; j++) {
put_bits(pb, s->extra_bits, s->predictor_buf[j][i]);
}
}
}
// apply lpc and entropy coding to audio samples
for (i = 0; i < 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
int32_t *residual = s->predictor_buf[i];
for (j = s->frame_size - 1; j > 0; j--)
residual[j] -= residual[j - 1];
}
alac_entropy_coder(s, i);
}
}
}
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
uint8_t * const *samples)
{
PutBitContext *pb = &s->pbctx;
const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
int ch, element, sce, cpe;
init_put_bits(pb, avpkt->data, avpkt->size);
ch = element = sce = cpe = 0;
while (ch < s->avctx->channels) {
if (ch_elements[element] == TYPE_CPE) {
write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
samples[ch_map[ch + 1]]);
cpe++;
ch += 2;
} else {
write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
sce++;
ch++;
}
element++;
}
put_bits(pb, 3, TYPE_END);
flush_put_bits(pb);
return put_bytes_output(pb);
}
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);
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_S32P) {
if (avctx->bits_per_raw_sample != 24)
av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
avctx->bits_per_raw_sample = 24;
} else {
avctx->bits_per_raw_sample = 16;
s->extra_bits = 0;
}
// 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,
avctx->bits_per_raw_sample);
avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + AV_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
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, avctx->bits_per_raw_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_raw_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);
}
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 AVERROR(EINVAL);
}
s->avctx = avctx;
if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
s->max_prediction_order,
FF_LPC_TYPE_LEVINSON)) < 0) {
return ret;
}
return 0;
}
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;
s->frame_size = frame->nb_samples;
if (frame->nb_samples < DEFAULT_FRAME_SIZE)
max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
avctx->bits_per_raw_sample);
else
max_frame_size = s->max_coded_frame_size;
if ((ret = ff_alloc_packet(avctx, avpkt, 4 * max_frame_size)) < 0)
return ret;
/* use verbatim mode for compression_level 0 */
if (s->compression_level) {
s->verbatim = 0;
s->extra_bits = avctx->bits_per_raw_sample - 16;
} else {
s->verbatim = 1;
s->extra_bits = 0;
}
out_bytes = write_frame(s, avpkt, frame->extended_data);
if (out_bytes > max_frame_size) {
/* frame too large. use verbatim mode */
s->verbatim = 1;
s->extra_bits = 0;
out_bytes = write_frame(s, avpkt, frame->extended_data);
}
avpkt->size = out_bytes;
*got_packet_ptr = 1;
return 0;
}
#define OFFSET(x) offsetof(AlacEncodeContext, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
{ "min_prediction_order", NULL, OFFSET(min_prediction_order), AV_OPT_TYPE_INT, { .i64 = DEFAULT_MIN_PRED_ORDER }, MIN_LPC_ORDER, ALAC_MAX_LPC_ORDER, AE },
{ "max_prediction_order", NULL, OFFSET(max_prediction_order), AV_OPT_TYPE_INT, { .i64 = DEFAULT_MAX_PRED_ORDER }, MIN_LPC_ORDER, ALAC_MAX_LPC_ORDER, AE },
{ NULL },
};
static const AVClass alacenc_class = {
.class_name = "alacenc",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
const AVCodec ff_alac_encoder = {
.name = "alac",
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ALAC,
.priv_data_size = sizeof(AlacEncodeContext),
.priv_class = &alacenc_class,
.init = alac_encode_init,
.encode2 = alac_encode_frame,
.close = alac_encode_close,
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME,
.channel_layouts = ff_alac_channel_layouts,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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