Commit 7bd3096f authored by Vladimir Voroshilov's avatar Vladimir Voroshilov Committed by Vitor Sessak
Browse files

SIPR decoder for modes 8k5, 6k5 and 5k0.

Patch by Vladimir Voroshilov and myself.

Originally committed as revision 21125 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 9ea97721
......@@ -48,6 +48,7 @@ version <next>:
- R210 decoder
- Auravision Aura 1 and 2 decoders
- Deluxe Paint Animation playback system
- SIPR decoding for modes 8k5, 6k5 and 5k0
......
......@@ -618,6 +618,8 @@ following image formats are supported:
@tab Real 28800 bit/s codec
@item RealAudio 3.0 (dnet) @tab IX @tab X
@tab Real low bitrate AC-3 codec
@item RealAudio SIPR / ACELP.NET @tab @tab X
@tab 16 kbps mode not yet supported
@item Shorten @tab @tab X
@item Sierra VMD audio @tab @tab X
@tab Used in Sierra VMD files.
......
......@@ -260,6 +260,9 @@ OBJS-$(CONFIG_RV40_DECODER) += rv40.o rv34.o h264pred.o rv40dsp.o \
OBJS-$(CONFIG_SGI_DECODER) += sgidec.o
OBJS-$(CONFIG_SGI_ENCODER) += sgienc.o rle.o
OBJS-$(CONFIG_SHORTEN_DECODER) += shorten.o
OBJS-$(CONFIG_SIPR_DECODER) += sipr.o acelp_pitch_delay.o \
celp_math.o acelp_vectors.o \
acelp_filters.o celp_filters.o lsp.o
OBJS-$(CONFIG_SMACKAUD_DECODER) += smacker.o
OBJS-$(CONFIG_SMACKER_DECODER) += smacker.o
OBJS-$(CONFIG_SMC_DECODER) += smc.o
......
......@@ -232,6 +232,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (RA_144, ra_144);
REGISTER_DECODER (RA_288, ra_288);
REGISTER_DECODER (SHORTEN, shorten);
REGISTER_DECODER (SIPR, sipr);
REGISTER_DECODER (SMACKAUD, smackaud);
REGISTER_ENCDEC (SONIC, sonic);
REGISTER_ENCODER (SONIC_LS, sonic_ls);
......
......@@ -30,7 +30,7 @@
#include "libavutil/avutil.h"
#define LIBAVCODEC_VERSION_MAJOR 52
#define LIBAVCODEC_VERSION_MINOR 46
#define LIBAVCODEC_VERSION_MINOR 47
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
......
/*
* SIPR / ACELP.NET decoder
*
* Copyright (c) 2008 Vladimir Voroshilov
* 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>
#include "avcodec.h"
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.h"
#include "lsp.h"
#include "celp_math.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "acelp_filters.h"
#include "celp_filters.h"
#define LSFQ_DIFF_MIN (0.0125 * M_PI)
#define LP_FILTER_ORDER 10
/** Number of past samples needed for excitation interpolation */
#define L_INTERPOL (LP_FILTER_ORDER + 1)
/** Subframe size for all modes except 16k */
#define SUBFR_SIZE 48
#include "siprdata.h"
typedef enum {
MODE_16k,
MODE_8k5,
MODE_6k5,
MODE_5k0,
MODE_COUNT
} SiprMode;
typedef struct {
const char *mode_name;
uint16_t bits_per_frame;
uint8_t subframe_count;
uint8_t frames_per_packet;
float pitch_sharp_factor;
/* bitstream parameters */
uint8_t number_of_fc_indexes;
/** size in bits of the i-th stage vector of quantizer */
uint8_t vq_indexes_bits[5];
/** size in bits of the adaptive-codebook index for every subframe */
uint8_t pitch_delay_bits[5];
uint8_t gp_index_bits;
uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes
} SiprModeParam;
static const SiprModeParam modes[MODE_COUNT] = {
[MODE_8k5] = {
.mode_name = "8k5",
.bits_per_frame = 152,
.subframe_count = 3,
.frames_per_packet = 1,
.pitch_sharp_factor = 0.8,
.number_of_fc_indexes = 3,
.vq_indexes_bits = {6, 7, 7, 7, 5},
.pitch_delay_bits = {8, 5, 5},
.gp_index_bits = 0,
.fc_index_bits = {9, 9, 9},
.gc_index_bits = 7
},
[MODE_6k5] = {
.mode_name = "6k5",
.bits_per_frame = 232,
.subframe_count = 3,
.frames_per_packet = 2,
.pitch_sharp_factor = 0.8,
.number_of_fc_indexes = 3,
.vq_indexes_bits = {6, 7, 7, 7, 5},
.pitch_delay_bits = {8, 5, 5},
.gp_index_bits = 0,
.fc_index_bits = {5, 5, 5},
.gc_index_bits = 7
},
[MODE_5k0] = {
.mode_name = "5k0",
.bits_per_frame = 296,
.subframe_count = 5,
.frames_per_packet = 2,
.pitch_sharp_factor = 0.85,
.number_of_fc_indexes = 1,
.vq_indexes_bits = {6, 7, 7, 7, 5},
.pitch_delay_bits = {8, 5, 8, 5, 5},
.gp_index_bits = 0,
.fc_index_bits = {10},
.gc_index_bits = 7
}
};
typedef struct {
AVCodecContext *avctx;
DSPContext dsp;
SiprModeParam m;
SiprMode mode;
float past_pitch_gain;
float lsf_history[LP_FILTER_ORDER];
float excitation[L_INTERPOL + PITCH_DELAY_MAX + 5*SUBFR_SIZE];
DECLARE_ALIGNED_16(float, synth_buf[LP_FILTER_ORDER + 5*SUBFR_SIZE + 6]);
float lsp_history[LP_FILTER_ORDER];
float gain_mem;
float energy_history[4];
float highpass_filt_mem[2];
float postfilter_mem[PITCH_DELAY_MAX + LP_FILTER_ORDER];
/* 5k0 */
float tilt_mem;
float postfilter_agc;
float postfilter_mem5k0[PITCH_DELAY_MAX + LP_FILTER_ORDER];
float postfilter_syn5k0[LP_FILTER_ORDER + SUBFR_SIZE*5];
} SiprContext;
typedef struct {
int vq_indexes[5];
int pitch_delay[5]; ///< pitch delay
int gp_index[5]; ///< adaptive-codebook gain indexes
int16_t fc_indexes[5][10]; ///< fixed-codebook indexes
int gc_index[5]; ///< fixed-codebook gain indexes
} SiprParameters;
static void dequant(float *out, const int *idx, const float *cbs[])
{
int i;
int stride = 2;
int num_vec = 5;
for (i = 0; i < num_vec; i++)
memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
}
static void lsf_decode_fp(float *lsfnew, float *lsf_history,
const SiprParameters *parm)
{
int i;
float lsf_tmp[LP_FILTER_ORDER];
dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
for (i = 0; i < LP_FILTER_ORDER; i++)
lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1);
/* Note that a minimum distance is not enforced between the last value and
the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1);
lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
for (i = 0; i < LP_FILTER_ORDER - 1; i++)
lsfnew[i] = cos(lsfnew[i]);
lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
}
/** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
static void pitch_sharpening(int pitch_lag_int, float beta,
float *fixed_vector)
{
int i;
for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
}
/**
* Extracts decoding parameters from the input bitstream.
* @param parms parameters structure
* @param pgb pointer to initialized GetBitContext structure
*/
static void decode_parameters(SiprParameters* parms, GetBitContext *pgb,
const SiprModeParam *p)
{
int i, j;
for (i = 0; i < 5; i++)
parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]);
for (i = 0; i < p->subframe_count; i++) {
parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]);
parms->gp_index[i] = get_bits(pgb, p->gp_index_bits);
for (j = 0; j < p->number_of_fc_indexes; j++)
parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
parms->gc_index[i] = get_bits(pgb, p->gc_index_bits);
}
}
static void lsp2lpc_sipr(const double *lsp, float *Az)
{
int lp_half_order = LP_FILTER_ORDER >> 1;
double buf[lp_half_order + 1];
double pa[lp_half_order + 1];
double *qa = buf + 1;
int i,j;
qa[-1] = 0.0;
ff_lsp2polyf(lsp , pa, lp_half_order );
ff_lsp2polyf(lsp + 1, qa, lp_half_order - 1);
for (i = 1, j = LP_FILTER_ORDER - 1; i < lp_half_order; i++, j--) {
double paf = pa[i] * (1 + lsp[LP_FILTER_ORDER - 1]);
double qaf = (qa[i] - qa[i-2]) * (1 - lsp[LP_FILTER_ORDER - 1]);
Az[i-1] = (paf + qaf) * 0.5;
Az[j-1] = (paf - qaf) * 0.5;
}
Az[lp_half_order - 1] = (1.0 + lsp[LP_FILTER_ORDER - 1]) *
pa[lp_half_order] * 0.5;
Az[LP_FILTER_ORDER - 1] = lsp[LP_FILTER_ORDER - 1];
}
static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
int num_subfr)
{
double lsfint[LP_FILTER_ORDER];
int i,j;
float t, t0 = 1.0 / num_subfr;
t = t0 * 0.5;
for (i = 0; i < num_subfr; i++) {
for (j = 0; j < LP_FILTER_ORDER; j++)
lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
lsp2lpc_sipr(lsfint, Az);
Az += LP_FILTER_ORDER;
t += t0;
}
}
/**
* Evaluates the adaptative impulse response.
*/
static void eval_ir(const float *Az, int pitch_lag, float *freq,
float pitch_sharp_factor)
{
float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
int i;
tmp1[0] = 1.;
for (i = 0; i < LP_FILTER_ORDER; i++) {
tmp1[i+1] = Az[i] * ff_pow_0_55[i];
tmp2[i ] = Az[i] * ff_pow_0_7 [i];
}
memset(tmp1 + 11, 0, 37 * sizeof(float));
ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE,
LP_FILTER_ORDER);
pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
}
/**
* Evaluates the convolution of a vector with a sparse vector.
*/
static void convolute_with_sparse(float *out, const AMRFixed *pulses,
const float *shape, int length)
{
int i, j;
memset(out, 0, length*sizeof(float));
for (i = 0; i < pulses->n; i++)
for (j = pulses->x[i]; j < length; j++)
out[j] += pulses->y[i] * shape[j - pulses->x[i]];
}
/**
* Apply postfilter, very similar to AMR one.
*/
static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
{
float buf[SUBFR_SIZE + LP_FILTER_ORDER];
float *pole_out = buf + LP_FILTER_ORDER;
float lpc_n[LP_FILTER_ORDER];
float lpc_d[LP_FILTER_ORDER];
int i;
for (i = 0; i < LP_FILTER_ORDER; i++) {
lpc_d[i] = lpc[i] * ff_pow_0_75[i];
lpc_n[i] = lpc[i] * pow_0_5 [i];
};
memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
LP_FILTER_ORDER*sizeof(float));
ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE,
LP_FILTER_ORDER);
memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
LP_FILTER_ORDER*sizeof(float));
ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
LP_FILTER_ORDER*sizeof(*pole_out));
memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
LP_FILTER_ORDER*sizeof(*pole_out));
ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE,
LP_FILTER_ORDER);
}
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
SiprMode mode, int low_gain)
{
int i;
switch (mode) {
case MODE_6k5:
for (i = 0; i < 3; i++) {
fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
}
fixed_sparse->n = 3;
break;
case MODE_8k5:
for (i = 0; i < 3; i++) {
fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i;
fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0;
fixed_sparse->y[2*i + 1] =
(fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
-fixed_sparse->y[2*i ] : fixed_sparse->y[2*i];
}
fixed_sparse->n = 6;
break;
case MODE_5k0:
default:
if (low_gain) {
int offset = (pulses[0] & 0x200) ? 2 : 0;
int val = pulses[0];
for (i = 0; i < 3; i++) {
int index = (val & 0x7) * 6 + 4 - i*2;
fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
fixed_sparse->x[i] = index;
val >>= 3;
}
fixed_sparse->n = 3;
} else {
int pulse_subset = (pulses[0] >> 8) & 1;
fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1;
fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
fixed_sparse->y[1] = -fixed_sparse->y[0];
fixed_sparse->n = 2;
}
break;
}
}
static void decode_frame(SiprContext *ctx, SiprParameters *params,
float *out_data)
{
int i, j;
int frame_size = ctx->m.subframe_count * SUBFR_SIZE;
float Az[LP_FILTER_ORDER * ctx->m.subframe_count];
float *excitation;
float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
float lsf_new[LP_FILTER_ORDER];
float *impulse_response = ir_buf + LP_FILTER_ORDER;
float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
// memory alignment
int t0_first = 0;
AMRFixed fixed_cb;
memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
lsf_decode_fp(lsf_new, ctx->lsf_history, params);
sipr_decode_lp(lsf_new, ctx->lsp_history, Az, ctx->m.subframe_count);
memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
for (i = 0; i < ctx->m.subframe_count; i++) {
float *pAz = Az + i*LP_FILTER_ORDER;
float fixed_vector[SUBFR_SIZE];
int T0,T0_frac;
float pitch_gain, gain_code, avg_energy;
ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
ctx->mode == MODE_5k0, 6);
if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
t0_first = T0;
ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
ff_b60_sinc, 6,
2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
SUBFR_SIZE);
decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
ctx->past_pitch_gain < 0.8);
eval_ir(pAz, T0, impulse_response, ctx->m.pitch_sharp_factor);
convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
SUBFR_SIZE);
avg_energy =
(0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/
SUBFR_SIZE;
ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
avg_energy, ctx->energy_history,
34 - 15.0/(log2f(10.0) * 0.05),
pred);
ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
pitch_gain, gain_code, SUBFR_SIZE);
pitch_gain *= 0.5 * pitch_gain;
pitch_gain = FFMIN(pitch_gain, 0.4);
ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
gain_code *= ctx->gain_mem;
for (j = 0; j < SUBFR_SIZE; j++)
fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
if (ctx->mode == MODE_5k0) {
postfilter_5k0(ctx, pAz, fixed_vector);
ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
pAz, excitation, SUBFR_SIZE,
LP_FILTER_ORDER);
}
ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
SUBFR_SIZE, LP_FILTER_ORDER);
excitation += SUBFR_SIZE;
}
memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
LP_FILTER_ORDER * sizeof(float));
if (ctx->mode == MODE_5k0) {
for (i = 0; i < ctx->m.subframe_count; i++) {
float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
SUBFR_SIZE);
ff_adaptative_gain_control(&synth[i * SUBFR_SIZE], energy,
SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
}
memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
LP_FILTER_ORDER*sizeof(float));
}
memcpy(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
(PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
ff_acelp_apply_order_2_transfer_function(synth,
(const float[2]) {-1.99997 , 1.000000000},
(const float[2]) {-1.93307352, 0.935891986},
0.939805806,
ctx->highpass_filt_mem,
frame_size);
ctx->dsp.vector_clipf(out_data, synth, -1, 32767./(1<<15), frame_size);
}
static av_cold int sipr_decoder_init(AVCodecContext * avctx)
{
SiprContext *ctx = avctx->priv_data;
int i;
if (avctx->bit_rate > 12200) ctx->mode = MODE_16k;
else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5;
else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5;
else ctx->mode = MODE_5k0;
ctx->m = modes[ctx->mode];
av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", ctx->m.mode_name);
for (i = 0; i < LP_FILTER_ORDER; i++)
ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
for (i = 0; i < 4; i++)
ctx->energy_history[i] = -14;
avctx->sample_fmt = SAMPLE_FMT_FLT;
if (ctx->mode == MODE_16k) {
av_log(avctx, AV_LOG_ERROR, "decoding 16kbps SIPR files is not "
"supported yet.\n");
return -1;
}
dsputil_init(&ctx->dsp, avctx);
return 0;
}
static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
int *data_size, AVPacket *avpkt)
{
SiprContext *ctx = avctx->priv_data;
const uint8_t *buf=avpkt->data;
SiprParameters parm;
GetBitContext gb;
float *data = datap;
int i;