mirror of
https://github.com/KolibriOS/kolibrios.git
synced 2024-12-15 19:33:59 +03:00
754f9336f0
git-svn-id: svn://kolibrios.org@4349 a494cfbc-eb01-0410-851d-a64ba20cac60
575 lines
19 KiB
C
575 lines
19 KiB
C
/*
|
|
* 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 <string.h>
|
|
|
|
#include "libavutil/channel_layout.h"
|
|
#include "libavutil/float_dsp.h"
|
|
#include "libavutil/mathematics.h"
|
|
#include "avcodec.h"
|
|
#define BITSTREAM_READER_LE
|
|
#include "get_bits.h"
|
|
#include "internal.h"
|
|
|
|
#include "lsp.h"
|
|
#include "acelp_vectors.h"
|
|
#include "acelp_pitch_delay.h"
|
|
#include "acelp_filters.h"
|
|
#include "celp_filters.h"
|
|
|
|
#define MAX_SUBFRAME_COUNT 5
|
|
|
|
#include "sipr.h"
|
|
#include "siprdata.h"
|
|
|
|
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;
|
|
uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor
|
|
|
|
/** 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_16k] = {
|
|
.mode_name = "16k",
|
|
.bits_per_frame = 160,
|
|
.subframe_count = SUBFRAME_COUNT_16k,
|
|
.frames_per_packet = 1,
|
|
.pitch_sharp_factor = 0.00,
|
|
|
|
.number_of_fc_indexes = 10,
|
|
.ma_predictor_bits = 1,
|
|
.vq_indexes_bits = {7, 8, 7, 7, 7},
|
|
.pitch_delay_bits = {9, 6},
|
|
.gp_index_bits = 4,
|
|
.fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
|
|
.gc_index_bits = 5
|
|
},
|
|
|
|
[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,
|
|
.ma_predictor_bits = 0,
|
|
.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,
|
|
.ma_predictor_bits = 0,
|
|
.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,
|
|
.ma_predictor_bits = 0,
|
|
.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
|
|
}
|
|
};
|
|
|
|
const float ff_pow_0_5[] = {
|
|
1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4),
|
|
1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8),
|
|
1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
|
|
1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
|
|
};
|
|
|
|
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];
|
|
}
|
|
|
|
/**
|
|
* Extract 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;
|
|
|
|
if (p->ma_predictor_bits)
|
|
parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits);
|
|
|
|
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]);
|
|
if (p->gp_index_bits)
|
|
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 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];
|
|
|
|
ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
|
|
Az += LP_FILTER_ORDER;
|
|
t += t0;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Evaluate the adaptive 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] = 1.0;
|
|
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);
|
|
}
|
|
|
|
/**
|
|
* Evaluate 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] * ff_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 subframe_count = modes[ctx->mode].subframe_count;
|
|
int frame_size = subframe_count * SUBFR_SIZE;
|
|
float Az[LP_FILTER_ORDER * MAX_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, 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 < 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, modes[ctx->mode].pitch_sharp_factor);
|
|
|
|
convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
|
|
SUBFR_SIZE);
|
|
|
|
avg_energy = (0.01 + avpriv_scalarproduct_float_c(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/(0.05*M_LN10/M_LN2),
|
|
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 < subframe_count; i++) {
|
|
float energy = avpriv_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
|
|
ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
|
|
SUBFR_SIZE);
|
|
ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
|
|
&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));
|
|
}
|
|
memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
|
|
(PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
|
|
|
|
ff_acelp_apply_order_2_transfer_function(out_data, synth,
|
|
(const float[2]) {-1.99997 , 1.000000000},
|
|
(const float[2]) {-1.93307352, 0.935891986},
|
|
0.939805806,
|
|
ctx->highpass_filt_mem,
|
|
frame_size);
|
|
}
|
|
|
|
static av_cold int sipr_decoder_init(AVCodecContext * avctx)
|
|
{
|
|
SiprContext *ctx = avctx->priv_data;
|
|
int i;
|
|
|
|
switch (avctx->block_align) {
|
|
case 20: ctx->mode = MODE_16k; break;
|
|
case 19: ctx->mode = MODE_8k5; break;
|
|
case 29: ctx->mode = MODE_6k5; break;
|
|
case 37: ctx->mode = MODE_5k0; break;
|
|
default:
|
|
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;
|
|
av_log(avctx, AV_LOG_WARNING,
|
|
"Invalid block_align: %d. Mode %s guessed based on bitrate: %d\n",
|
|
avctx->block_align, modes[ctx->mode].mode_name, avctx->bit_rate);
|
|
}
|
|
|
|
av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
|
|
|
|
if (ctx->mode == MODE_16k) {
|
|
ff_sipr_init_16k(ctx);
|
|
ctx->decode_frame = ff_sipr_decode_frame_16k;
|
|
} else {
|
|
ctx->decode_frame = decode_frame;
|
|
}
|
|
|
|
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->channels = 1;
|
|
avctx->channel_layout = AV_CH_LAYOUT_MONO;
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int sipr_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
SiprContext *ctx = avctx->priv_data;
|
|
AVFrame *frame = data;
|
|
const uint8_t *buf=avpkt->data;
|
|
SiprParameters parm;
|
|
const SiprModeParam *mode_par = &modes[ctx->mode];
|
|
GetBitContext gb;
|
|
float *samples;
|
|
int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
|
|
int i, ret;
|
|
|
|
ctx->avctx = avctx;
|
|
if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"Error processing packet: packet size (%d) too small\n",
|
|
avpkt->size);
|
|
return -1;
|
|
}
|
|
|
|
/* get output buffer */
|
|
frame->nb_samples = mode_par->frames_per_packet * subframe_size *
|
|
mode_par->subframe_count;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
samples = (float *)frame->data[0];
|
|
|
|
init_get_bits(&gb, buf, mode_par->bits_per_frame);
|
|
|
|
for (i = 0; i < mode_par->frames_per_packet; i++) {
|
|
decode_parameters(&parm, &gb, mode_par);
|
|
|
|
ctx->decode_frame(ctx, &parm, samples);
|
|
|
|
samples += subframe_size * mode_par->subframe_count;
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return mode_par->bits_per_frame >> 3;
|
|
}
|
|
|
|
AVCodec ff_sipr_decoder = {
|
|
.name = "sipr",
|
|
.long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_SIPR,
|
|
.priv_data_size = sizeof(SiprContext),
|
|
.init = sipr_decoder_init,
|
|
.decode = sipr_decode_frame,
|
|
.capabilities = CODEC_CAP_DR1,
|
|
};
|