kolibrios/contrib/sdk/sources/ffmpeg/ffmpeg-2.1/libavcodec/metasound.c
Sergey Semyonov (Serge) ecf3e862ea ffmpeg-2.1.1: move directory
git-svn-id: svn://kolibrios.org@6148 a494cfbc-eb01-0410-851d-a64ba20cac60
2016-02-05 22:14:10 +00:00

344 lines
12 KiB
C

/*
* Voxware MetaSound decoder
* Copyright (c) 2013 Konstantin Shishkov
* based on TwinVQ decoder
* Copyright (c) 2009 Vitor Sessak
*
* 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 <math.h>
#include <stdint.h>
#define BITSTREAM_READER_LE
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "get_bits.h"
#include "fft.h"
#include "internal.h"
#include "lsp.h"
#include "sinewin.h"
#include "twinvq.h"
#include "metasound_data.h"
static void add_peak(float period, int width, const float *shape,
float ppc_gain, float *speech, int len)
{
int i, j, center;
const float *shape_end = shape + len;
// First peak centered around zero
for (i = 0; i < width / 2; i++)
speech[i] += ppc_gain * *shape++;
for (i = 1; i < ROUNDED_DIV(len, width); i++) {
center = (int)(i * period + 0.5);
for (j = -width / 2; j < (width + 1) / 2; j++)
speech[j + center] += ppc_gain * *shape++;
}
// For the last block, be careful not to go beyond the end of the buffer
center = (int)(i * period + 0.5);
for (j = -width / 2; j < (width + 1) / 2 && shape < shape_end; j++)
speech[j + center] += ppc_gain * *shape++;
}
static void decode_ppc(TwinVQContext *tctx, int period_coef, int g_coef,
const float *shape, float *speech)
{
const TwinVQModeTab *mtab = tctx->mtab;
int isampf = tctx->avctx->sample_rate / 1000;
int ibps = tctx->avctx->bit_rate / (1000 * tctx->avctx->channels);
int width;
float ratio = (float)mtab->size / isampf;
float min_period, max_period, period_range, period;
float some_mult;
float pgain_base, pgain_step, ppc_gain;
if (tctx->avctx->channels == 1) {
min_period = log2(ratio * 0.2);
max_period = min_period + log2(6);
} else {
min_period = (int)(ratio * 0.2 * 400 + 0.5) / 400.0;
max_period = (int)(ratio * 0.2 * 400 * 6 + 0.5) / 400.0;
}
period_range = max_period - min_period;
period = min_period + period_coef * period_range /
((1 << mtab->ppc_period_bit) - 1);
if (tctx->avctx->channels == 1)
period = powf(2.0, period);
else
period = (int)(period * 400 + 0.5) / 400.0;
switch (isampf) {
case 8: some_mult = 2.0; break;
case 11: some_mult = 3.0; break;
case 16: some_mult = 3.0; break;
case 22: some_mult = ibps == 32 ? 2.0 : 4.0; break;
case 44: some_mult = 8.0; break;
default: some_mult = 4.0;
}
width = (int)(some_mult / (mtab->size / period) * mtab->ppc_shape_len);
if (isampf == 22 && ibps == 32)
width = (int)((2.0 / period + 1) * width + 0.5);
pgain_base = tctx->avctx->channels == 2 ? 25000.0 : 20000.0;
pgain_step = pgain_base / ((1 << mtab->pgain_bit) - 1);
ppc_gain = 1.0 / 8192 *
twinvq_mulawinv(pgain_step * g_coef + pgain_step / 2,
pgain_base, TWINVQ_PGAIN_MU);
add_peak(period, width, shape, ppc_gain, speech, mtab->ppc_shape_len);
}
static void dec_bark_env(TwinVQContext *tctx, const uint8_t *in, int use_hist,
int ch, float *out, float gain,
enum TwinVQFrameType ftype)
{
const TwinVQModeTab *mtab = tctx->mtab;
int i, j;
float *hist = tctx->bark_hist[ftype][ch];
float val = ((const float []) { 0.4, 0.35, 0.28 })[ftype];
int bark_n_coef = mtab->fmode[ftype].bark_n_coef;
int fw_cb_len = mtab->fmode[ftype].bark_env_size / bark_n_coef;
int idx = 0;
if (tctx->avctx->channels == 1)
val = 0.5;
for (i = 0; i < fw_cb_len; i++)
for (j = 0; j < bark_n_coef; j++, idx++) {
float tmp2 = mtab->fmode[ftype].bark_cb[fw_cb_len * in[j] + i] *
(1.0 / 2048);
float st;
if (tctx->avctx->channels == 1)
st = use_hist ?
tmp2 + val * hist[idx] + 1.0 : tmp2 + 1.0;
else
st = use_hist ? (1.0 - val) * tmp2 + val * hist[idx] + 1.0
: tmp2 + 1.0;
hist[idx] = tmp2;
if (st < 0.1)
st = 0.1;
twinvq_memset_float(out, st * gain,
mtab->fmode[ftype].bark_tab[idx]);
out += mtab->fmode[ftype].bark_tab[idx];
}
}
static void read_cb_data(TwinVQContext *tctx, GetBitContext *gb,
uint8_t *dst, enum TwinVQFrameType ftype)
{
int i;
for (i = 0; i < tctx->n_div[ftype]; i++) {
int bs_second_part = (i >= tctx->bits_main_spec_change[ftype]);
*dst++ = get_bits(gb, tctx->bits_main_spec[0][ftype][bs_second_part]);
*dst++ = get_bits(gb, tctx->bits_main_spec[1][ftype][bs_second_part]);
}
}
static int metasound_read_bitstream(AVCodecContext *avctx, TwinVQContext *tctx,
const uint8_t *buf, int buf_size)
{
TwinVQFrameData *bits = &tctx->bits;
const TwinVQModeTab *mtab = tctx->mtab;
int channels = tctx->avctx->channels;
int sub;
GetBitContext gb;
int i, j, k;
init_get_bits(&gb, buf, buf_size * 8);
bits->window_type = get_bits(&gb, TWINVQ_WINDOW_TYPE_BITS);
if (bits->window_type > 8) {
av_log(avctx, AV_LOG_ERROR, "Invalid window type, broken sample?\n");
return AVERROR_INVALIDDATA;
}
bits->ftype = ff_twinvq_wtype_to_ftype_table[tctx->bits.window_type];
sub = mtab->fmode[bits->ftype].sub;
if (bits->ftype != TWINVQ_FT_SHORT)
get_bits(&gb, 2);
read_cb_data(tctx, &gb, bits->main_coeffs, bits->ftype);
for (i = 0; i < channels; i++)
for (j = 0; j < sub; j++)
for (k = 0; k < mtab->fmode[bits->ftype].bark_n_coef; k++)
bits->bark1[i][j][k] =
get_bits(&gb, mtab->fmode[bits->ftype].bark_n_bit);
for (i = 0; i < channels; i++)
for (j = 0; j < sub; j++)
bits->bark_use_hist[i][j] = get_bits1(&gb);
if (bits->ftype == TWINVQ_FT_LONG) {
for (i = 0; i < channels; i++)
bits->gain_bits[i] = get_bits(&gb, TWINVQ_GAIN_BITS);
} else {
for (i = 0; i < channels; i++) {
bits->gain_bits[i] = get_bits(&gb, TWINVQ_GAIN_BITS);
for (j = 0; j < sub; j++)
bits->sub_gain_bits[i * sub + j] =
get_bits(&gb, TWINVQ_SUB_GAIN_BITS);
}
}
for (i = 0; i < channels; i++) {
bits->lpc_hist_idx[i] = get_bits(&gb, mtab->lsp_bit0);
bits->lpc_idx1[i] = get_bits(&gb, mtab->lsp_bit1);
for (j = 0; j < mtab->lsp_split; j++)
bits->lpc_idx2[i][j] = get_bits(&gb, mtab->lsp_bit2);
}
if (bits->ftype == TWINVQ_FT_LONG) {
read_cb_data(tctx, &gb, bits->ppc_coeffs, 3);
for (i = 0; i < channels; i++) {
bits->p_coef[i] = get_bits(&gb, mtab->ppc_period_bit);
bits->g_coef[i] = get_bits(&gb, mtab->pgain_bit);
}
}
return (get_bits_count(&gb) + 7) / 8;
}
typedef struct MetasoundProps {
uint32_t tag;
int bit_rate;
int channels;
int sample_rate;
} MetasoundProps;
static const MetasoundProps codec_props[] = {
{ MKTAG('V','X','0','3'), 6, 1, 8000 },
{ MKTAG('V','X','0','4'), 12, 2, 8000 },
{ MKTAG('V','O','X','i'), 8, 1, 8000 },
{ MKTAG('V','O','X','j'), 10, 1, 11025 },
{ MKTAG('V','O','X','k'), 16, 1, 16000 },
{ MKTAG('V','O','X','L'), 24, 1, 22050 },
{ MKTAG('V','O','X','q'), 32, 1, 44100 },
{ MKTAG('V','O','X','r'), 40, 1, 44100 },
{ MKTAG('V','O','X','s'), 48, 1, 44100 },
{ MKTAG('V','O','X','t'), 16, 2, 8000 },
{ MKTAG('V','O','X','u'), 20, 2, 11025 },
{ MKTAG('V','O','X','v'), 32, 2, 16000 },
{ MKTAG('V','O','X','w'), 48, 2, 22050 },
{ MKTAG('V','O','X','x'), 64, 2, 44100 },
{ MKTAG('V','O','X','y'), 80, 2, 44100 },
{ MKTAG('V','O','X','z'), 96, 2, 44100 },
{ 0, 0, 0, 0 }
};
static av_cold int metasound_decode_init(AVCodecContext *avctx)
{
int isampf, ibps;
TwinVQContext *tctx = avctx->priv_data;
uint32_t tag;
const MetasoundProps *props = codec_props;
if (!avctx->extradata || avctx->extradata_size < 16) {
av_log(avctx, AV_LOG_ERROR, "Missing or incomplete extradata\n");
return AVERROR_INVALIDDATA;
}
tag = AV_RL32(avctx->extradata + 12);
for (;;) {
if (!props->tag) {
av_log(avctx, AV_LOG_ERROR, "Could not find tag %08X\n", tag);
return AVERROR_INVALIDDATA;
}
if (props->tag == tag) {
avctx->sample_rate = props->sample_rate;
avctx->channels = props->channels;
avctx->bit_rate = props->bit_rate * 1000;
isampf = avctx->sample_rate / 1000;
break;
}
props++;
}
if (avctx->channels <= 0 || avctx->channels > TWINVQ_CHANNELS_MAX) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %i\n",
avctx->channels);
return AVERROR_INVALIDDATA;
}
avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO
: AV_CH_LAYOUT_STEREO;
ibps = avctx->bit_rate / (1000 * avctx->channels);
switch ((avctx->channels << 16) + (isampf << 8) + ibps) {
case (1 << 16) + ( 8 << 8) + 8:
tctx->mtab = &ff_metasound_mode0808;
break;
case (1 << 16) + (16 << 8) + 16:
tctx->mtab = &ff_metasound_mode1616;
break;
case (1 << 16) + (44 << 8) + 32:
tctx->mtab = &ff_metasound_mode4432;
break;
case (2 << 16) + (44 << 8) + 48:
tctx->mtab = &ff_metasound_mode4448s;
break;
default:
av_log(avctx, AV_LOG_ERROR,
"This version does not support %d kHz - %d kbit/s/ch mode.\n",
isampf, isampf);
return AVERROR(ENOSYS);
}
avctx->block_align = (avctx->bit_rate * tctx->mtab->size
/ avctx->sample_rate + 7) / 8;
tctx->codec = TWINVQ_CODEC_METASOUND;
tctx->read_bitstream = metasound_read_bitstream;
tctx->dec_bark_env = dec_bark_env;
tctx->decode_ppc = decode_ppc;
return ff_twinvq_decode_init(avctx);
}
AVCodec ff_metasound_decoder = {
.name = "metasound",
.long_name = NULL_IF_CONFIG_SMALL("Voxware MetaSound"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_METASOUND,
.priv_data_size = sizeof(TwinVQContext),
.init = metasound_decode_init,
.close = ff_twinvq_decode_close,
.decode = ff_twinvq_decode_frame,
.capabilities = CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};