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/*
* 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> |
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#include <string.h> |
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|
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#include "libavutil/channel_layout.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/mathematics.h" |
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#include "avcodec.h" |
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#define BITSTREAM_READER_LE |
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#include "get_bits.h" |
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#include "internal.h" |
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#include "lsp.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "acelp_filters.h"
#include "celp_filters.h"
|
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#define MAX_SUBFRAME_COUNT 5
|
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#include "sipr.h" |
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#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; |
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uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor |
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/** 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] = { |
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[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
},
|
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[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, |
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.ma_predictor_bits = 0, |
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.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, |
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.ma_predictor_bits = 0, |
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.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, |
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.ma_predictor_bits = 0, |
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.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
}
};
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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)
};
|
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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];
}
/** |
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* Extract decoding parameters from the input bitstream. |
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* @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;
|
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if (p->ma_predictor_bits)
parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits); |
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|
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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]); |
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if (p->gp_index_bits)
parms->gp_index[i] = get_bits(pgb, p->gp_index_bits); |
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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];
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ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER); |
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Az += LP_FILTER_ORDER;
t += t0;
}
}
/** |
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* Evaluate the adaptive impulse response. |
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*/
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;
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tmp1[0] = 1.0; |
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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);
}
/** |
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* Evaluate the convolution of a vector with a sparse vector. |
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*/
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]; |
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lpc_n[i] = lpc[i] * ff_pow_0_5 [i]; |
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};
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; |
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int subframe_count = modes[ctx->mode].subframe_count;
int frame_size = subframe_count * SUBFR_SIZE; |
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float Az[LP_FILTER_ORDER * MAX_SUBFRAME_COUNT]; |
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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);
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sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count); |
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memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
|
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for (i = 0; i < subframe_count; i++) { |
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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);
|
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eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor); |
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convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
SUBFR_SIZE);
|
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avg_energy = (0.01 + avpriv_scalarproduct_float_c(fixed_vector,
fixed_vector,
SUBFR_SIZE)) /
SUBFR_SIZE; |
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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, |
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34 - 15.0/(0.05*M_LN10/M_LN2), |
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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) { |
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for (i = 0; i < subframe_count; i++) { |
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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); |
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ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
&synth[i * SUBFR_SIZE], energy, |
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SUBFR_SIZE, 0.9, &ctx->postfilter_agc); |
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}
memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
LP_FILTER_ORDER*sizeof(float));
} |
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memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL, |
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(PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
|
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ff_acelp_apply_order_2_transfer_function(out_data, synth, |
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(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;
|
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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: |
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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; |
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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); |
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} |
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|
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av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name); |
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|
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if (ctx->mode == MODE_16k) { |
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ff_sipr_init_16k(ctx); |
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ctx->decode_frame = ff_sipr_decode_frame_16k;
} else {
ctx->decode_frame = decode_frame;
} |
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|
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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;
|
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avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
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return 0;
}
|
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static int sipr_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt) |
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{
SiprContext *ctx = avctx->priv_data; |
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AVFrame *frame = data; |
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const uint8_t *buf=avpkt->data;
SiprParameters parm; |
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const SiprModeParam *mode_par = &modes[ctx->mode]; |
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GetBitContext gb; |
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float *samples; |
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int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE; |
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int i, ret; |
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ctx->avctx = avctx; |
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if (avpkt->size < (mode_par->bits_per_frame >> 3)) { |
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av_log(avctx, AV_LOG_ERROR,
"Error processing packet: packet size (%d) too small\n",
avpkt->size);
return -1;
} |
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|
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/* get output buffer */ |
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frame->nb_samples = mode_par->frames_per_packet * subframe_size *
mode_par->subframe_count; |
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
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return ret; |
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samples = (float *)frame->data[0]; |
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|
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init_get_bits(&gb, buf, mode_par->bits_per_frame); |
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|
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for (i = 0; i < mode_par->frames_per_packet; i++) {
decode_parameters(&parm, &gb, mode_par); |
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|
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ctx->decode_frame(ctx, &parm, samples); |
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|
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samples += subframe_size * mode_par->subframe_count; |
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}
|
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*got_frame_ptr = 1; |
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|
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return mode_par->bits_per_frame >> 3; |
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} |
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|
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AVCodec ff_sipr_decoder = { |
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.name = "sipr", |
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.long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_SIPR, |
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.priv_data_size = sizeof(SiprContext),
.init = sipr_decoder_init,
.decode = sipr_decode_frame, |
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.capabilities = CODEC_CAP_DR1, |
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}; |