ffmpeg/libavcodec/wmaenc.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

461 lines
15 KiB
C

/*
* WMA compatible encoder
* Copyright (c) 2007 Michael Niedermayer
*
* 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/attributes.h"
#include "libavutil/ffmath.h"
#include "avcodec.h"
#include "encode.h"
#include "internal.h"
#include "wma.h"
#include "libavutil/avassert.h"
static av_cold int encode_init(AVCodecContext *avctx)
{
WMACodecContext *s = avctx->priv_data;
int i, flags1, flags2, block_align;
uint8_t *extradata;
int ret;
s->avctx = avctx;
if (avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR,
"too many channels: got %i, need %i or fewer\n",
avctx->channels, MAX_CHANNELS);
return AVERROR(EINVAL);
}
if (avctx->sample_rate > 48000) {
av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz\n",
avctx->sample_rate);
return AVERROR(EINVAL);
}
if (avctx->bit_rate < 24 * 1000) {
av_log(avctx, AV_LOG_ERROR,
"bitrate too low: got %"PRId64", need 24000 or higher\n",
avctx->bit_rate);
return AVERROR(EINVAL);
}
/* extract flag info */
flags1 = 0;
flags2 = 1;
if (avctx->codec->id == AV_CODEC_ID_WMAV1) {
extradata = av_malloc(4);
if (!extradata)
return AVERROR(ENOMEM);
avctx->extradata_size = 4;
AV_WL16(extradata, flags1);
AV_WL16(extradata + 2, flags2);
} else if (avctx->codec->id == AV_CODEC_ID_WMAV2) {
extradata = av_mallocz(10);
if (!extradata)
return AVERROR(ENOMEM);
avctx->extradata_size = 10;
AV_WL32(extradata, flags1);
AV_WL16(extradata + 4, flags2);
} else {
av_assert0(0);
}
avctx->extradata = extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
if (avctx->channels == 2)
s->ms_stereo = 1;
if ((ret = ff_wma_init(avctx, flags2)) < 0)
return ret;
/* init MDCT */
for (i = 0; i < s->nb_block_sizes; i++) {
ret = ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
if (ret < 0)
return ret;
}
block_align = avctx->bit_rate * (int64_t) s->frame_len /
(avctx->sample_rate * 8);
block_align = FFMIN(block_align, MAX_CODED_SUPERFRAME_SIZE);
avctx->block_align = block_align;
avctx->frame_size = avctx->initial_padding = s->frame_len;
return 0;
}
static int apply_window_and_mdct(AVCodecContext *avctx, const AVFrame *frame)
{
WMACodecContext *s = avctx->priv_data;
float **audio = (float **) frame->extended_data;
int len = frame->nb_samples;
int window_index = s->frame_len_bits - s->block_len_bits;
FFTContext *mdct = &s->mdct_ctx[window_index];
int ch;
const float *win = s->windows[window_index];
int window_len = 1 << s->block_len_bits;
float n = 2.0 * 32768.0 / window_len;
for (ch = 0; ch < avctx->channels; ch++) {
memcpy(s->output, s->frame_out[ch], window_len * sizeof(*s->output));
s->fdsp->vector_fmul_scalar(s->frame_out[ch], audio[ch], n, len);
s->fdsp->vector_fmul_reverse(&s->output[window_len], s->frame_out[ch],
win, len);
s->fdsp->vector_fmul(s->frame_out[ch], s->frame_out[ch], win, len);
mdct->mdct_calc(mdct, s->coefs[ch], s->output);
if (!isfinite(s->coefs[ch][0])) {
av_log(avctx, AV_LOG_ERROR, "Input contains NaN/+-Inf\n");
return AVERROR(EINVAL);
}
}
return 0;
}
// FIXME use for decoding too
static void init_exp(WMACodecContext *s, int ch, const int *exp_param)
{
int n;
const uint16_t *ptr;
float v, *q, max_scale, *q_end;
ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
q = s->exponents[ch];
q_end = q + s->block_len;
max_scale = 0;
while (q < q_end) {
/* XXX: use a table */
v = ff_exp10(*exp_param++ *(1.0 / 16.0));
max_scale = FFMAX(max_scale, v);
n = *ptr++;
do {
*q++ = v;
} while (--n);
}
s->max_exponent[ch] = max_scale;
}
static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param)
{
int last_exp;
const uint16_t *ptr;
float *q, *q_end;
ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
q = s->exponents[ch];
q_end = q + s->block_len;
if (s->version == 1) {
last_exp = *exp_param++;
av_assert0(last_exp - 10 >= 0 && last_exp - 10 < 32);
put_bits(&s->pb, 5, last_exp - 10);
q += *ptr++;
} else
last_exp = 36;
while (q < q_end) {
int exp = *exp_param++;
int code = exp - last_exp + 60;
av_assert1(code >= 0 && code < 120);
put_bits(&s->pb, ff_aac_scalefactor_bits[code],
ff_aac_scalefactor_code[code]);
/* XXX: use a table */
q += *ptr++;
last_exp = exp;
}
}
static int encode_block(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
int total_gain)
{
int v, bsize, ch, coef_nb_bits, parse_exponents;
float mdct_norm;
int nb_coefs[MAX_CHANNELS];
static const int fixed_exp[25] = {
20, 20, 20, 20, 20,
20, 20, 20, 20, 20,
20, 20, 20, 20, 20,
20, 20, 20, 20, 20,
20, 20, 20, 20, 20
};
// FIXME remove duplication relative to decoder
if (s->use_variable_block_len) {
av_assert0(0); // FIXME not implemented
} else {
/* fixed block len */
s->next_block_len_bits = s->frame_len_bits;
s->prev_block_len_bits = s->frame_len_bits;
s->block_len_bits = s->frame_len_bits;
}
s->block_len = 1 << s->block_len_bits;
// av_assert0((s->block_pos + s->block_len) <= s->frame_len);
bsize = s->frame_len_bits - s->block_len_bits;
// FIXME factor
v = s->coefs_end[bsize] - s->coefs_start;
for (ch = 0; ch < s->avctx->channels; ch++)
nb_coefs[ch] = v;
{
int n4 = s->block_len / 2;
mdct_norm = 1.0 / (float) n4;
if (s->version == 1)
mdct_norm *= sqrt(n4);
}
if (s->avctx->channels == 2)
put_bits(&s->pb, 1, !!s->ms_stereo);
for (ch = 0; ch < s->avctx->channels; ch++) {
// FIXME only set channel_coded when needed, instead of always
s->channel_coded[ch] = 1;
if (s->channel_coded[ch])
init_exp(s, ch, fixed_exp);
}
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
WMACoef *coefs1;
float *coefs, *exponents, mult;
int i, n;
coefs1 = s->coefs1[ch];
exponents = s->exponents[ch];
mult = ff_exp10(total_gain * 0.05) / s->max_exponent[ch];
mult *= mdct_norm;
coefs = src_coefs[ch];
if (s->use_noise_coding && 0) {
av_assert0(0); // FIXME not implemented
} else {
coefs += s->coefs_start;
n = nb_coefs[ch];
for (i = 0; i < n; i++) {
double t = *coefs++ / (exponents[i] * mult);
if (t < -32768 || t > 32767)
return -1;
coefs1[i] = lrint(t);
}
}
}
}
v = 0;
for (ch = 0; ch < s->avctx->channels; ch++) {
int a = s->channel_coded[ch];
put_bits(&s->pb, 1, a);
v |= a;
}
if (!v)
return 1;
for (v = total_gain - 1; v >= 127; v -= 127)
put_bits(&s->pb, 7, 127);
put_bits(&s->pb, 7, v);
coef_nb_bits = ff_wma_total_gain_to_bits(total_gain);
if (s->use_noise_coding) {
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int i, n;
n = s->exponent_high_sizes[bsize];
for (i = 0; i < n; i++) {
put_bits(&s->pb, 1, s->high_band_coded[ch][i] = 0);
if (0)
nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
}
}
}
}
parse_exponents = 1;
if (s->block_len_bits != s->frame_len_bits)
put_bits(&s->pb, 1, parse_exponents);
if (parse_exponents) {
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
if (s->use_exp_vlc) {
encode_exp_vlc(s, ch, fixed_exp);
} else {
av_assert0(0); // FIXME not implemented
// encode_exp_lsp(s, ch);
}
}
}
} else
av_assert0(0); // FIXME not implemented
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int run, tindex;
WMACoef *ptr, *eptr;
tindex = (ch == 1 && s->ms_stereo);
ptr = &s->coefs1[ch][0];
eptr = ptr + nb_coefs[ch];
run = 0;
for (; ptr < eptr; ptr++) {
if (*ptr) {
int level = *ptr;
int abs_level = FFABS(level);
int code = 0;
if (abs_level <= s->coef_vlcs[tindex]->max_level)
if (run < s->coef_vlcs[tindex]->levels[abs_level - 1])
code = run + s->int_table[tindex][abs_level - 1];
av_assert2(code < s->coef_vlcs[tindex]->n);
put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[code],
s->coef_vlcs[tindex]->huffcodes[code]);
if (code == 0) {
if (1 << coef_nb_bits <= abs_level)
return -1;
put_bits(&s->pb, coef_nb_bits, abs_level);
put_bits(&s->pb, s->frame_len_bits, run);
}
// FIXME the sign is flipped somewhere
put_bits(&s->pb, 1, level < 0);
run = 0;
} else
run++;
}
if (run)
put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[1],
s->coef_vlcs[tindex]->huffcodes[1]);
}
if (s->version == 1 && s->avctx->channels >= 2)
align_put_bits(&s->pb);
}
return 0;
}
static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
uint8_t *buf, int buf_size, int total_gain)
{
init_put_bits(&s->pb, buf, buf_size);
if (s->use_bit_reservoir)
av_assert0(0); // FIXME not implemented
else if (encode_block(s, src_coefs, total_gain) < 0)
return INT_MAX;
align_put_bits(&s->pb);
return put_bits_count(&s->pb) / 8 - s->avctx->block_align;
}
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data;
int i, total_gain, ret, error;
s->block_len_bits = s->frame_len_bits; // required by non variable block len
s->block_len = 1 << s->block_len_bits;
ret = apply_window_and_mdct(avctx, frame);
if (ret < 0)
return ret;
if (s->ms_stereo) {
float a, b;
int i;
for (i = 0; i < s->block_len; i++) {
a = s->coefs[0][i] * 0.5;
b = s->coefs[1][i] * 0.5;
s->coefs[0][i] = a + b;
s->coefs[1][i] = a - b;
}
}
if ((ret = ff_alloc_packet(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE)) < 0)
return ret;
total_gain = 128;
for (i = 64; i; i >>= 1) {
error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if (error <= 0)
total_gain -= i;
}
while(total_gain <= 128 && error > 0)
error = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain++);
if (error > 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid input data or requested bitrate too low, cannot encode\n");
avpkt->size = 0;
return AVERROR(EINVAL);
}
av_assert0((put_bits_count(&s->pb) & 7) == 0);
i = avctx->block_align - put_bytes_count(&s->pb, 0);
av_assert0(i>=0);
while(i--)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == avctx->block_align);
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
avpkt->size = avctx->block_align;
*got_packet_ptr = 1;
return 0;
}
#if CONFIG_WMAV1_ENCODER
const AVCodec ff_wmav1_encoder = {
.name = "wmav1",
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};
#endif
#if CONFIG_WMAV2_ENCODER
const AVCodec ff_wmav2_encoder = {
.name = "wmav2",
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};
#endif