Commit d31ba231 authored by Francesco Lavra's avatar Francesco Lavra Committed by Vitor Sessak
Browse files

RealAudio 14.4k encoder.

Patch by Francesco Lavra (firstnamelastname@interfree.it)

Originally committed as revision 23579 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent b6c265ec
......@@ -89,6 +89,7 @@ version 0.6:
- 35% faster VP3/Theora decoding
- faster AAC decoding
- faster H.264 decoding
- RealAudio 1.0 (14.4K) encoder
......
......@@ -1270,6 +1270,7 @@ png_decoder_select="zlib"
png_encoder_select="zlib"
qcelp_decoder_select="lsp"
qdm2_decoder_select="mdct rdft"
real_144_encoder_select="lpc"
rv10_decoder_select="h263_decoder"
rv10_encoder_select="h263_encoder"
rv20_decoder_select="h263_decoder"
......
......@@ -635,7 +635,7 @@ following image formats are supported:
@item QCELP / PureVoice @tab @tab X
@item QDesign Music Codec 2 @tab @tab X
@tab There are still some distortions.
@item RealAudio 1.0 (14.4K) @tab @tab X
@item RealAudio 1.0 (14.4K) @tab X @tab X
@tab Real 14400 bit/s codec
@item RealAudio 2.0 (28.8K) @tab @tab X
@tab Real 28800 bit/s codec
......
......@@ -282,6 +282,7 @@ OBJS-$(CONFIG_QTRLE_DECODER) += qtrle.o
OBJS-$(CONFIG_QTRLE_ENCODER) += qtrleenc.o
OBJS-$(CONFIG_R210_DECODER) += r210dec.o
OBJS-$(CONFIG_RA_144_DECODER) += ra144dec.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_144_ENCODER) += ra144enc.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_288_DECODER) += ra288.o celp_math.o celp_filters.o
OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o
OBJS-$(CONFIG_RAWVIDEO_ENCODER) += rawenc.o
......
......@@ -247,7 +247,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (NELLYMOSER, nellymoser);
REGISTER_DECODER (QCELP, qcelp);
REGISTER_DECODER (QDM2, qdm2);
REGISTER_DECODER (RA_144, ra_144);
REGISTER_ENCDEC (RA_144, ra_144);
REGISTER_DECODER (RA_288, ra_288);
REGISTER_DECODER (SHORTEN, shorten);
REGISTER_DECODER (SIPR, sipr);
......
......@@ -30,8 +30,8 @@
#include "libavutil/avutil.h"
#define LIBAVCODEC_VERSION_MAJOR 52
#define LIBAVCODEC_VERSION_MINOR 75
#define LIBAVCODEC_VERSION_MICRO 1
#define LIBAVCODEC_VERSION_MINOR 76
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
......
......@@ -23,13 +23,18 @@
#define AVCODEC_RA144_H
#include <stdint.h>
#include "dsputil.h"
#define NBLOCKS 4 ///< number of subblocks within a block
#define BLOCKSIZE 40 ///< subblock size in 16-bit words
#define BUFFERSIZE 146 ///< the size of the adaptive codebook
#define FIXED_CB_SIZE 128 ///< size of fixed codebooks
#define FRAMESIZE 20 ///< size of encoded frame
#define LPC_ORDER 10 ///< order of LPC filter
typedef struct {
AVCodecContext *avctx;
DSPContext dsp;
unsigned int old_energy; ///< previous frame energy
......@@ -41,6 +46,8 @@ typedef struct {
unsigned int lpc_refl_rms[2];
int16_t curr_block[NBLOCKS * BLOCKSIZE];
/** The current subblock padded by the last 10 values of the previous one. */
int16_t curr_sblock[50];
......
/*
* Real Audio 1.0 (14.4K) encoder
* Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
*
* 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
*/
/**
* @file libavcodec/ra144enc.c
* Real Audio 1.0 (14.4K) encoder
* @author Francesco Lavra <francescolavra@interfree.it>
*/
#include <values.h>
#include "avcodec.h"
#include "put_bits.h"
#include "lpc.h"
#include "celp_filters.h"
#include "ra144.h"
static av_cold int ra144_encode_init(AVCodecContext * avctx)
{
RA144Context *ractx;
if (avctx->sample_fmt != SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "invalid sample format\n");
return -1;
}
if (avctx->channels != 1) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
avctx->channels);
return -1;
}
avctx->frame_size = NBLOCKS * BLOCKSIZE;
avctx->bit_rate = 8000;
ractx = avctx->priv_data;
ractx->lpc_coef[0] = ractx->lpc_tables[0];
ractx->lpc_coef[1] = ractx->lpc_tables[1];
ractx->avctx = avctx;
dsputil_init(&ractx->dsp, avctx);
return 0;
}
/**
* Quantizes a value by searching a sorted table for the element with the
* nearest value
*
* @param value value to quantize
* @param table array containing the quantization table
* @param size size of the quantization table
* @return index of the quantization table corresponding to the element with the
* nearest value
*/
static int quantize(int value, const int16_t *table, unsigned int size)
{
unsigned int low = 0, high = size - 1;
while (1) {
int index = (low + high) >> 1;
int error = table[index] - value;
if (index == low)
return table[high] + error > value ? low : high;
if (error > 0) {
high = index;
} else {
low = index;
}
}
}
/**
* Orthogonalizes a vector to another vector
*
* @param v vector to orthogonalize
* @param u vector against which orthogonalization is performed
*/
static void orthogonalize(float *v, const float *u)
{
int i;
float num = 0, den = 0;
for (i = 0; i < BLOCKSIZE; i++) {
num += v[i] * u[i];
den += u[i] * u[i];
}
num /= den;
for (i = 0; i < BLOCKSIZE; i++)
v[i] -= num * u[i];
}
/**
* Calculates match score and gain of an LPC-filtered vector with respect to
* input data, possibly othogonalizing it to up to 2 other vectors
*
* @param work array used to calculate the filtered vector
* @param coefs coefficients of the LPC filter
* @param vect original vector
* @param ortho1 first vector against which orthogonalization is performed
* @param ortho2 second vector against which orthogonalization is performed
* @param data input data
* @param score pointer to variable where match score is returned
* @param gain pointer to variable where gain is returned
*/
static void get_match_score(float *work, const float *coefs, float *vect,
const float *ortho1, const float *ortho2,
const float *data, float *score, float *gain)
{
float c, g;
int i;
ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
if (ortho1)
orthogonalize(work, ortho1);
if (ortho2)
orthogonalize(work, ortho2);
c = g = 0;
for (i = 0; i < BLOCKSIZE; i++) {
g += work[i] * work[i];
c += data[i] * work[i];
}
if (c <= 0) {
*score = 0;
return;
}
*gain = c / g;
*score = *gain * c;
}
/**
* Creates a vector from the adaptive codebook at a given lag value
*
* @param vect array where vector is stored
* @param cb adaptive codebook
* @param lag lag value
*/
static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
{
int i;
cb += BUFFERSIZE - lag;
for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)
vect[i] = cb[i];
if (lag < BLOCKSIZE)
for (i = 0; i < BLOCKSIZE - lag; i++)
vect[lag + i] = cb[i];
}
/**
* Searches the adaptive codebook for the best entry and gain and removes its
* contribution from input data
*
* @param adapt_cb array from which the adaptive codebook is extracted
* @param work array used to calculate LPC-filtered vectors
* @param coefs coefficients of the LPC filter
* @param data input data
* @return index of the best entry of the adaptive codebook
*/
static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
const float *coefs, float *data)
{
int i, best_vect;
float score, gain, best_score, best_gain;
float exc[BLOCKSIZE];
gain = best_score = 0;
for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {
create_adapt_vect(exc, adapt_cb, i);
get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);
if (score > best_score) {
best_score = score;
best_vect = i;
best_gain = gain;
}
}
if (!best_score)
return 0;
/**
* Re-calculate the filtered vector from the vector with maximum match score
* and remove its contribution from input data.
*/
create_adapt_vect(exc, adapt_cb, best_vect);
ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);
for (i = 0; i < BLOCKSIZE; i++)
data[i] -= best_gain * work[i];
return (best_vect - BLOCKSIZE / 2 + 1);
}
/**
* Finds the best vector of a fixed codebook by applying an LPC filter to
* codebook entries, possibly othogonalizing them to up to 2 other vectors and
* matching the results with input data
*
* @param work array used to calculate the filtered vectors
* @param coefs coefficients of the LPC filter
* @param cb fixed codebook
* @param ortho1 first vector against which orthogonalization is performed
* @param ortho2 second vector against which orthogonalization is performed
* @param data input data
* @param idx pointer to variable where the index of the best codebook entry is
* returned
* @param gain pointer to variable where the gain of the best codebook entry is
* returned
*/
static void find_best_vect(float *work, const float *coefs,
const int8_t cb[][BLOCKSIZE], const float *ortho1,
const float *ortho2, float *data, int *idx,
float *gain)
{
int i, j;
float g, score, best_score;
float vect[BLOCKSIZE];
*idx = *gain = best_score = 0;
for (i = 0; i < FIXED_CB_SIZE; i++) {
for (j = 0; j < BLOCKSIZE; j++)
vect[j] = cb[i][j];
get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);
if (score > best_score) {
best_score = score;
*idx = i;
*gain = g;
}
}
}
/**
* Searches the two fixed codebooks for the best entry and gain
*
* @param work array used to calculate LPC-filtered vectors
* @param coefs coefficients of the LPC filter
* @param data input data
* @param cba_idx index of the best entry of the adaptive codebook
* @param cb1_idx pointer to variable where the index of the best entry of the
* first fixed codebook is returned
* @param cb2_idx pointer to variable where the index of the best entry of the
* second fixed codebook is returned
*/
static void fixed_cb_search(float *work, const float *coefs, float *data,
int cba_idx, int *cb1_idx, int *cb2_idx)
{
int i, ortho_cb1;
float gain;
float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];
float vect[BLOCKSIZE];
/**
* The filtered vector from the adaptive codebook can be retrieved from
* work, because this function is called just after adaptive_cb_search().
*/
if (cba_idx)
memcpy(cba_vect, work, sizeof(cba_vect));
find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,
data, cb1_idx, &gain);
/**
* Re-calculate the filtered vector from the vector with maximum match score
* and remove its contribution from input data.
*/
if (gain) {
for (i = 0; i < BLOCKSIZE; i++)
vect[i] = ff_cb1_vects[*cb1_idx][i];
ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
if (cba_idx)
orthogonalize(work, cba_vect);
for (i = 0; i < BLOCKSIZE; i++)
data[i] -= gain * work[i];
memcpy(cb1_vect, work, sizeof(cb1_vect));
ortho_cb1 = 1;
} else
ortho_cb1 = 0;
find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,
ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
}
/**
* Encodes a subblock of the current frame
*
* @param ractx encoder context
* @param sblock_data input data of the subblock
* @param lpc_coefs coefficients of the LPC filter
* @param rms RMS of the reflection coefficients
* @param pb pointer to PutBitContext of the current frame
*/
static void ra144_encode_subblock(RA144Context *ractx,
const int16_t *sblock_data,
const int16_t *lpc_coefs, unsigned int rms,
PutBitContext *pb)
{
float data[BLOCKSIZE], work[LPC_ORDER + BLOCKSIZE];
float coefs[LPC_ORDER];
float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
int16_t cba_vect[BLOCKSIZE];
int cba_idx, cb1_idx, cb2_idx, gain;
int i, n, m[3];
float g[3];
float error, best_error;
for (i = 0; i < LPC_ORDER; i++) {
work[i] = ractx->curr_sblock[BLOCKSIZE + i];
coefs[i] = lpc_coefs[i] * (1/4096.0);
}
/**
* Calculate the zero-input response of the LPC filter and subtract it from
* input data.
*/
memset(data, 0, sizeof(data));
ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,
LPC_ORDER);
for (i = 0; i < BLOCKSIZE; i++) {
zero[i] = work[LPC_ORDER + i];
data[i] = sblock_data[i] - zero[i];
}
/**
* Codebook search is performed without taking into account the contribution
* of the previous subblock, since it has been just subtracted from input
* data.
*/
memset(work, 0, LPC_ORDER * sizeof(*work));
cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,
data);
if (cba_idx) {
/**
* The filtered vector from the adaptive codebook can be retrieved from
* work, see implementation of adaptive_cb_search().
*/
memcpy(cba, work + LPC_ORDER, sizeof(cba));
ff_copy_and_dup(cba_vect, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
m[0] = (ff_irms(cba_vect) * rms) >> 12;
}
fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
for (i = 0; i < BLOCKSIZE; i++) {
cb1[i] = ff_cb1_vects[cb1_idx][i];
cb2[i] = ff_cb2_vects[cb2_idx][i];
}
ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE,
LPC_ORDER);
memcpy(cb1, work + LPC_ORDER, sizeof(cb1));
m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;
ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE,
LPC_ORDER);
memcpy(cb2, work + LPC_ORDER, sizeof(cb2));
m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;
best_error = FLT_MAX;
gain = 0;
for (n = 0; n < 256; n++) {
g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *
(1/4096.0);
g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *
(1/4096.0);
error = 0;
if (cba_idx) {
g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *
(1/4096.0);
for (i = 0; i < BLOCKSIZE; i++) {
data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
g[2] * cb2[i];
error += (data[i] - sblock_data[i]) *
(data[i] - sblock_data[i]);
}
} else {
for (i = 0; i < BLOCKSIZE; i++) {
data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
error += (data[i] - sblock_data[i]) *
(data[i] - sblock_data[i]);
}
}
if (error < best_error) {
best_error = error;
gain = n;
}
}
put_bits(pb, 7, cba_idx);
put_bits(pb, 8, gain);
put_bits(pb, 7, cb1_idx);
put_bits(pb, 7, cb2_idx);
ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,
gain);
}
static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
{
static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
RA144Context *ractx;
PutBitContext pb;
int32_t lpc_data[NBLOCKS * BLOCKSIZE];
int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
int shift[LPC_ORDER];
int16_t block_coefs[NBLOCKS][LPC_ORDER];
int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */
unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
int energy = 0;
int i, idx;
if (buf_size < FRAMESIZE) {
av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
return 0;
}
ractx = avctx->priv_data;
/**
* Since the LPC coefficients are calculated on a frame centered over the
* fourth subframe, to encode a given frame, data from the next frame is
* needed. In each call to this function, the previous frame (whose data are
* saved in the encoder context) is encoded, and data from the current frame
* are saved in the encoder context to be used in the next function call.
*/
for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
energy += (lpc_data[i] * lpc_data[i]) >> 4;
}
for (i = 2 * BLOCKSIZE + BLOCKSIZE / 2; i < NBLOCKS * BLOCKSIZE; i++) {
lpc_data[i] = *((int16_t *)data + i - 2 * BLOCKSIZE - BLOCKSIZE / 2) >>
2;
energy += (lpc_data[i] * lpc_data[i]) >> 4;
}
energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
32)];
ff_lpc_calc_coefs(&ractx->dsp, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
LPC_ORDER, 16, lpc_coefs, shift, 1, ORDER_METHOD_EST, 12,
0);
for (i = 0; i < LPC_ORDER; i++)
block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<
(12 - shift[LPC_ORDER - 1]));
/**
* TODO: apply perceptual weighting of the input speech through bandwidth
* expansion of the LPC filter.
*/
if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
/**
* The filter is unstable: use the coefficients of the previous frame.
*/
ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx);
}
init_put_bits(&pb, frame, buf_size);
for (i = 0; i < LPC_ORDER; i++) {
idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
put_bits(&pb, bit_sizes[i], idx);
lpc_refl[i] = ff_lpc_refl_cb[i][idx];
}
ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
energy <= ractx->old_energy,
ff_t_sqrt(energy * ractx->old_energy) >> 12);
refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);
put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));
for (i = 0; i < NBLOCKS; i++)
ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,
block_coefs[i], refl_rms[i], &pb);
flush_put_bits(&pb);
ractx->old_energy = energy;
ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
for (i = 0; i < NBLOCKS * BLOCKSIZE; i++)
ractx->curr_block[i] = *((int16_t *)data + i) >> 2;
return FRAMESIZE;
}
AVCodec ra_144_encoder =
{
"real_144",
CODEC_TYPE_AUDIO,
CODEC_ID_RA_144,
sizeof(RA144Context),
ra144_encode_init,
ra144_encode_frame,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K) encoder"),
};
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