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FFmpeg/libavcodec/sipr.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs 
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:15 -03:00

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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>
#include <string.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mathematics.h"
#define BITSTREAM_READER_LE
#include "avcodec.h"
#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 SiprModeParam {
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 * const 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: %"PRId64"\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 AVERROR_INVALIDDATA;
}
/* 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;
}
const 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 = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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