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FFmpeg/libavcodec/amrwbdec.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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/*
* AMR wideband decoder
* Copyright (c) 2010 Marcelo Galvao Povoa
*
* 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
* AMR wideband decoder
*/
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "lsp.h"
#include "celp_filters.h"
#include "celp_math.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "internal.h"
#define AMR_USE_16BIT_TABLES
#include "amr.h"
#include "amrwbdata.h"
#include "mips/amrwbdec_mips.h"
typedef struct AMRWBContext {
AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
enum Mode fr_cur_mode; ///< mode index of current frame
uint8_t fr_quality; ///< frame quality index (FQI)
float isf_cur[LP_ORDER]; ///< working ISF vector from current frame
float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
double isp[4][LP_ORDER]; ///< ISP vectors from current frame
double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe
float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
float *excitation; ///< points to current excitation in excitation_buf[]
float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz
float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling
float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
float demph_mem[1]; ///< previous value in the de-emphasis filter
float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
AVLFG prng; ///< random number generator for white noise excitation
uint8_t first_frame; ///< flag active during decoding of the first frame
ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
CELPMContext celpm_ctx; ///< context for fixed point math operations
} AMRWBContext;
static av_cold int amrwb_decode_init(AVCodecContext *avctx)
{
AMRWBContext *ctx = avctx->priv_data;
int i;
if (avctx->channels > 1) {
avpriv_report_missing_feature(avctx, "multi-channel AMR");
return AVERROR_PATCHWELCOME;
}
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
if (!avctx->sample_rate)
avctx->sample_rate = 16000;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
av_lfg_init(&ctx->prng, 1);
ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
ctx->first_frame = 1;
for (i = 0; i < LP_ORDER; i++)
ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
for (i = 0; i < 4; i++)
ctx->prediction_error[i] = MIN_ENERGY;
ff_acelp_filter_init(&ctx->acelpf_ctx);
ff_acelp_vectors_init(&ctx->acelpv_ctx);
ff_celp_filter_init(&ctx->celpf_ctx);
ff_celp_math_init(&ctx->celpm_ctx);
return 0;
}
/**
* Decode the frame header in the "MIME/storage" format. This format
* is simpler and does not carry the auxiliary frame information.
*
* @param[in] ctx The Context
* @param[in] buf Pointer to the input buffer
*
* @return The decoded header length in bytes
*/
static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
{
/* Decode frame header (1st octet) */
ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
ctx->fr_quality = (buf[0] & 0x4) == 0x4;
return 1;
}
/**
* Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
*
* @param[in] ind Array of 5 indexes
* @param[out] isf_q Buffer for isf_q[LP_ORDER]
*/
static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
{
int i;
for (i = 0; i < 9; i++)
isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
for (i = 0; i < 7; i++)
isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
for (i = 0; i < 5; i++)
isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
for (i = 0; i < 4; i++)
isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
for (i = 0; i < 7; i++)
isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
}
/**
* Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
*
* @param[in] ind Array of 7 indexes
* @param[out] isf_q Buffer for isf_q[LP_ORDER]
*/
static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
{
int i;
for (i = 0; i < 9; i++)
isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
for (i = 0; i < 7; i++)
isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
for (i = 0; i < 3; i++)
isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
for (i = 0; i < 3; i++)
isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
for (i = 0; i < 3; i++)
isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
for (i = 0; i < 3; i++)
isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
for (i = 0; i < 4; i++)
isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
}
/**
* Apply mean and past ISF values using the prediction factor.
* Updates past ISF vector.
*
* @param[in,out] isf_q Current quantized ISF
* @param[in,out] isf_past Past quantized ISF
*/
static void isf_add_mean_and_past(float *isf_q, float *isf_past)
{
int i;
float tmp;
for (i = 0; i < LP_ORDER; i++) {
tmp = isf_q[i];
isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
isf_q[i] += PRED_FACTOR * isf_past[i];
isf_past[i] = tmp;
}
}
/**
* Interpolate the fourth ISP vector from current and past frames
* to obtain an ISP vector for each subframe.
*
* @param[in,out] isp_q ISPs for each subframe
* @param[in] isp4_past Past ISP for subframe 4
*/
static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
{
int i, k;
for (k = 0; k < 3; k++) {
float c = isfp_inter[k];
for (i = 0; i < LP_ORDER; i++)
isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
}
}
/**
* Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
* Calculate integer lag and fractional lag always using 1/4 resolution.
* In 1st and 3rd subframes the index is relative to last subframe integer lag.
*
* @param[out] lag_int Decoded integer pitch lag
* @param[out] lag_frac Decoded fractional pitch lag
* @param[in] pitch_index Adaptive codebook pitch index
* @param[in,out] base_lag_int Base integer lag used in relative subframes
* @param[in] subframe Current subframe index (0 to 3)
*/
static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
uint8_t *base_lag_int, int subframe)
{
if (subframe == 0 || subframe == 2) {
if (pitch_index < 376) {
*lag_int = (pitch_index + 137) >> 2;
*lag_frac = pitch_index - (*lag_int << 2) + 136;
} else if (pitch_index < 440) {
*lag_int = (pitch_index + 257 - 376) >> 1;
*lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
/* the actual resolution is 1/2 but expressed as 1/4 */
} else {
*lag_int = pitch_index - 280;
*lag_frac = 0;
}
/* minimum lag for next subframe */
*base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
// XXX: the spec states clearly that *base_lag_int should be
// the nearest integer to *lag_int (minus 8), but the ref code
// actually always uses its floor, I'm following the latter
} else {
*lag_int = (pitch_index + 1) >> 2;
*lag_frac = pitch_index - (*lag_int << 2);
*lag_int += *base_lag_int;
}
}
/**
* Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
* The description is analogous to decode_pitch_lag_high, but in 6k60 the
* relative index is used for all subframes except the first.
*/
static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
uint8_t *base_lag_int, int subframe, enum Mode mode)
{
if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
if (pitch_index < 116) {
*lag_int = (pitch_index + 69) >> 1;
*lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
} else {
*lag_int = pitch_index - 24;
*lag_frac = 0;
}
// XXX: same problem as before
*base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
} else {
*lag_int = (pitch_index + 1) >> 1;
*lag_frac = (pitch_index - (*lag_int << 1)) * 2;
*lag_int += *base_lag_int;
}
}
/**
* Find the pitch vector by interpolating the past excitation at the
* pitch delay, which is obtained in this function.
*
* @param[in,out] ctx The context
* @param[in] amr_subframe Current subframe data
* @param[in] subframe Current subframe index (0 to 3)
*/
static void decode_pitch_vector(AMRWBContext *ctx,
const AMRWBSubFrame *amr_subframe,
const int subframe)
{
int pitch_lag_int, pitch_lag_frac;
int i;
float *exc = ctx->excitation;
enum Mode mode = ctx->fr_cur_mode;
if (mode <= MODE_8k85) {
decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
&ctx->base_pitch_lag, subframe, mode);
} else
decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
&ctx->base_pitch_lag, subframe);
ctx->pitch_lag_int = pitch_lag_int;
pitch_lag_int += pitch_lag_frac > 0;
/* Calculate the pitch vector by interpolating the past excitation at the
pitch lag using a hamming windowed sinc function */
ctx->acelpf_ctx.acelp_interpolatef(exc,
exc + 1 - pitch_lag_int,
ac_inter, 4,
pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
LP_ORDER, AMRWB_SFR_SIZE + 1);
/* Check which pitch signal path should be used
* 6k60 and 8k85 modes have the ltp flag set to 0 */
if (amr_subframe->ltp) {
memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
} else {
for (i = 0; i < AMRWB_SFR_SIZE; i++)
ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
0.18 * exc[i + 1];
memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
}
}
/** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
#define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len))
/** Get the bit at specified position */
#define BIT_POS(x, p) (((x) >> (p)) & 1)
/**
* The next six functions decode_[i]p_track decode exactly i pulses
* positions and amplitudes (-1 or 1) in a subframe track using
* an encoded pulse indexing (TS 26.190 section 5.8.2).
*
* The results are given in out[], in which a negative number means
* amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
*
* @param[out] out Output buffer (writes i elements)
* @param[in] code Pulse index (no. of bits varies, see below)
* @param[in] m (log2) Number of potential positions
* @param[in] off Offset for decoded positions
*/
static inline void decode_1p_track(int *out, int code, int m, int off)
{
int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
out[0] = BIT_POS(code, m) ? -pos : pos;
}
static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
{
int pos0 = BIT_STR(code, m, m) + off;
int pos1 = BIT_STR(code, 0, m) + off;
out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
out[1] = pos0 > pos1 ? -out[1] : out[1];
}
static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
{
int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
m - 1, off + half_2p);
decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
}
static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
{
int half_4p, subhalf_2p;
int b_offset = 1 << (m - 1);
switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
m - 2, off + half_4p + subhalf_2p);
decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
m - 1, off + half_4p);
break;
case 1: /* 1 pulse in A, 3 pulses in B */
decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
m - 1, off);
decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
m - 1, off + b_offset);
break;
case 2: /* 2 pulses in each half */
decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
m - 1, off);
decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
m - 1, off + b_offset);
break;
case 3: /* 3 pulses in A, 1 pulse in B */
decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
m - 1, off);
decode_1p_track(out + 3, BIT_STR(code, 0, m),
m - 1, off + b_offset);
break;
}
}
static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
{
int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
m - 1, off + half_3p);
decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
}
static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
{
int b_offset = 1 << (m - 1);
/* which half has more pulses in cases 0 to 2 */
int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
int half_other = b_offset - half_more;
switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
decode_1p_track(out, BIT_STR(code, 0, m),
m - 1, off + half_more);
decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
m - 1, off + half_more);
break;
case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
decode_1p_track(out, BIT_STR(code, 0, m),
m - 1, off + half_other);
decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
m - 1, off + half_more);
break;
case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
m - 1, off + half_other);
decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
m - 1, off + half_more);
break;
case 3: /* 3 pulses in A, 3 pulses in B */
decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
m - 1, off);
decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
m - 1, off + b_offset);
break;
}
}
/**
* Decode the algebraic codebook index to pulse positions and signs,
* then construct the algebraic codebook vector.
*
* @param[out] fixed_vector Buffer for the fixed codebook excitation
* @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
* @param[in] pulse_lo LSBs part of the pulse index array
* @param[in] mode Mode of the current frame
*/
static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
const uint16_t *pulse_lo, const enum Mode mode)
{
/* sig_pos stores for each track the decoded pulse position indexes
* (1-based) multiplied by its corresponding amplitude (+1 or -1) */
int sig_pos[4][6];
int spacing = (mode == MODE_6k60) ? 2 : 4;
int i, j;
switch (mode) {
case MODE_6k60:
for (i = 0; i < 2; i++)
decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
break;
case MODE_8k85:
for (i = 0; i < 4; i++)
decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
break;
case MODE_12k65:
for (i = 0; i < 4; i++)
decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
break;
case MODE_14k25:
for (i = 0; i < 2; i++)
decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
for (i = 2; i < 4; i++)
decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
break;
case MODE_15k85:
for (i = 0; i < 4; i++)
decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
break;
case MODE_18k25:
for (i = 0; i < 4; i++)
decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
((int) pulse_hi[i] << 14), 4, 1);
break;
case MODE_19k85:
for (i = 0; i < 2; i++)
decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
((int) pulse_hi[i] << 10), 4, 1);
for (i = 2; i < 4; i++)
decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
((int) pulse_hi[i] << 14), 4, 1);
break;
case MODE_23k05:
case MODE_23k85:
for (i = 0; i < 4; i++)
decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
((int) pulse_hi[i] << 11), 4, 1);
break;
}
memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
for (i = 0; i < 4; i++)
for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
}
}
/**
* Decode pitch gain and fixed gain correction factor.
*
* @param[in] vq_gain Vector-quantized index for gains
* @param[in] mode Mode of the current frame
* @param[out] fixed_gain_factor Decoded fixed gain correction factor
* @param[out] pitch_gain Decoded pitch gain
*/
static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
float *fixed_gain_factor, float *pitch_gain)
{
const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
qua_gain_7b[vq_gain]);
*pitch_gain = gains[0] * (1.0f / (1 << 14));
*fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
}
/**
* Apply pitch sharpening filters to the fixed codebook vector.
*
* @param[in] ctx The context
* @param[in,out] fixed_vector Fixed codebook excitation
*/
// XXX: Spec states this procedure should be applied when the pitch
// lag is less than 64, but this checking seems absent in reference and AMR-NB
static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
{
int i;
/* Tilt part */
for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
/* Periodicity enhancement part */
for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
}
/**
* Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
*
* @param[in] p_vector, f_vector Pitch and fixed excitation vectors
* @param[in] p_gain, f_gain Pitch and fixed gains
* @param[in] ctx The context
*/
// XXX: There is something wrong with the precision here! The magnitudes
// of the energies are not correct. Please check the reference code carefully
static float voice_factor(float *p_vector, float p_gain,
float *f_vector, float f_gain,
CELPMContext *ctx)
{
double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
AMRWB_SFR_SIZE) *
p_gain * p_gain;
double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
AMRWB_SFR_SIZE) *
f_gain * f_gain;
return (p_ener - f_ener) / (p_ener + f_ener + 0.01);
}
/**
* Reduce fixed vector sparseness by smoothing with one of three IR filters,
* also known as "adaptive phase dispersion".
*
* @param[in] ctx The context
* @param[in,out] fixed_vector Unfiltered fixed vector
* @param[out] buf Space for modified vector if necessary
*
* @return The potentially overwritten filtered fixed vector address
*/
static float *anti_sparseness(AMRWBContext *ctx,
float *fixed_vector, float *buf)
{
int ir_filter_nr;
if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
return fixed_vector;
if (ctx->pitch_gain[0] < 0.6) {
ir_filter_nr = 0; // strong filtering
} else if (ctx->pitch_gain[0] < 0.9) {
ir_filter_nr = 1; // medium filtering
} else
ir_filter_nr = 2; // no filtering
/* detect 'onset' */
if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
if (ir_filter_nr < 2)
ir_filter_nr++;
} else {
int i, count = 0;
for (i = 0; i < 6; i++)
if (ctx->pitch_gain[i] < 0.6)
count++;
if (count > 2)
ir_filter_nr = 0;
if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
ir_filter_nr--;
}
/* update ir filter strength history */
ctx->prev_ir_filter_nr = ir_filter_nr;
ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
if (ir_filter_nr < 2) {
int i;
const float *coef = ir_filters_lookup[ir_filter_nr];
/* Circular convolution code in the reference
* decoder was modified to avoid using one
* extra array. The filtered vector is given by:
*
* c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
*/
memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
for (i = 0; i < AMRWB_SFR_SIZE; i++)
if (fixed_vector[i])
ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
AMRWB_SFR_SIZE);
fixed_vector = buf;
}
return fixed_vector;
}
/**
* Calculate a stability factor {teta} based on distance between
* current and past isf. A value of 1 shows maximum signal stability.
*/
static float stability_factor(const float *isf, const float *isf_past)
{
int i;
float acc = 0.0;
for (i = 0; i < LP_ORDER - 1; i++)
acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
// XXX: This part is not so clear from the reference code
// the result is more accurate changing the "/ 256" to "* 512"
return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
}
/**
* Apply a non-linear fixed gain smoothing in order to reduce
* fluctuation in the energy of excitation.
*
* @param[in] fixed_gain Unsmoothed fixed gain
* @param[in,out] prev_tr_gain Previous threshold gain (updated)
* @param[in] voice_fac Frame voicing factor
* @param[in] stab_fac Frame stability factor
*
* @return The smoothed gain
*/
static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
float voice_fac, float stab_fac)
{
float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
float g0;
// XXX: the following fixed-point constants used to in(de)crement
// gain by 1.5dB were taken from the reference code, maybe it could
// be simpler
if (fixed_gain < *prev_tr_gain) {
g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
(6226 * (1.0f / (1 << 15)))); // +1.5 dB
} else
g0 = FFMAX(*prev_tr_gain, fixed_gain *
(27536 * (1.0f / (1 << 15)))); // -1.5 dB
*prev_tr_gain = g0; // update next frame threshold
return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
}
/**
* Filter the fixed_vector to emphasize the higher frequencies.
*
* @param[in,out] fixed_vector Fixed codebook vector
* @param[in] voice_fac Frame voicing factor
*/
static void pitch_enhancer(float *fixed_vector, float voice_fac)
{
int i;
float cpe = 0.125 * (1 + voice_fac);
float last = fixed_vector[0]; // holds c(i - 1)
fixed_vector[0] -= cpe * fixed_vector[1];
for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
float cur = fixed_vector[i];
fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
last = cur;
}
fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
}
/**
* Conduct 16th order linear predictive coding synthesis from excitation.
*
* @param[in] ctx Pointer to the AMRWBContext
* @param[in] lpc Pointer to the LPC coefficients
* @param[out] excitation Buffer for synthesis final excitation
* @param[in] fixed_gain Fixed codebook gain for synthesis
* @param[in] fixed_vector Algebraic codebook vector
* @param[in,out] samples Pointer to the output samples and memory
*/
static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
float fixed_gain, const float *fixed_vector,
float *samples)
{
ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
/* emphasize pitch vector contribution in low bitrate modes */
if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
int i;
float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
AMRWB_SFR_SIZE);
// XXX: Weird part in both ref code and spec. A unknown parameter
// {beta} seems to be identical to the current pitch gain
float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
for (i = 0; i < AMRWB_SFR_SIZE; i++)
excitation[i] += pitch_factor * ctx->pitch_vector[i];
ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
energy, AMRWB_SFR_SIZE);
}
ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
AMRWB_SFR_SIZE, LP_ORDER);
}
/**
* Apply to synthesis a de-emphasis filter of the form:
* H(z) = 1 / (1 - m * z^-1)
*
* @param[out] out Output buffer
* @param[in] in Input samples array with in[-1]
* @param[in] m Filter coefficient
* @param[in,out] mem State from last filtering
*/
static void de_emphasis(float *out, float *in, float m, float mem[1])
{
int i;
out[0] = in[0] + m * mem[0];
for (i = 1; i < AMRWB_SFR_SIZE; i++)
out[i] = in[i] + out[i - 1] * m;
mem[0] = out[AMRWB_SFR_SIZE - 1];
}
/**
* Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
* a FIR interpolation filter. Uses past data from before *in address.
*
* @param[out] out Buffer for interpolated signal
* @param[in] in Current signal data (length 0.8*o_size)
* @param[in] o_size Output signal length
* @param[in] ctx The context
*/
static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
{
const float *in0 = in - UPS_FIR_SIZE + 1;
int i, j, k;
int int_part = 0, frac_part;
i = 0;
for (j = 0; j < o_size / 5; j++) {
out[i] = in[int_part];
frac_part = 4;
i++;
for (k = 1; k < 5; k++) {
out[i] = ctx->dot_productf(in0 + int_part,
upsample_fir[4 - frac_part],
UPS_MEM_SIZE);
int_part++;
frac_part--;
i++;
}
}
}
/**
* Calculate the high-band gain based on encoded index (23k85 mode) or
* on the low-band speech signal and the Voice Activity Detection flag.
*
* @param[in] ctx The context
* @param[in] synth LB speech synthesis at 12.8k
* @param[in] hb_idx Gain index for mode 23k85 only
* @param[in] vad VAD flag for the frame
*/
static float find_hb_gain(AMRWBContext *ctx, const float *synth,
uint16_t hb_idx, uint8_t vad)
{
int wsp = (vad > 0);
float tilt;
float tmp;
if (ctx->fr_cur_mode == MODE_23k85)
return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1);
if (tmp > 0) {
tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
} else
tilt = 0;
/* return gain bounded by [0.1, 1.0] */
return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
}
/**
* Generate the high-band excitation with the same energy from the lower
* one and scaled by the given gain.
*
* @param[in] ctx The context
* @param[out] hb_exc Buffer for the excitation
* @param[in] synth_exc Low-band excitation used for synthesis
* @param[in] hb_gain Wanted excitation gain
*/
static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
const float *synth_exc, float hb_gain)
{
int i;
float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
AMRWB_SFR_SIZE);
/* Generate a white-noise excitation */
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
energy * hb_gain * hb_gain,
AMRWB_SFR_SIZE_16k);
}
/**
* Calculate the auto-correlation for the ISF difference vector.
*/
static float auto_correlation(float *diff_isf, float mean, int lag)
{
int i;
float sum = 0.0;
for (i = 7; i < LP_ORDER - 2; i++) {
float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
sum += prod * prod;
}
return sum;
}
/**
* Extrapolate a ISF vector to the 16kHz range (20th order LP)
* used at mode 6k60 LP filter for the high frequency band.
*
* @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
* values on input
*/
static void extrapolate_isf(float isf[LP_ORDER_16k])
{
float diff_isf[LP_ORDER - 2], diff_mean;
float corr_lag[3];
float est, scale;
int i, j, i_max_corr;
isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
/* Calculate the difference vector */
for (i = 0; i < LP_ORDER - 2; i++)
diff_isf[i] = isf[i + 1] - isf[i];
diff_mean = 0.0;
for (i = 2; i < LP_ORDER - 2; i++)
diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
/* Find which is the maximum autocorrelation */
i_max_corr = 0;
for (i = 0; i < 3; i++) {
corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
if (corr_lag[i] > corr_lag[i_max_corr])
i_max_corr = i;
}
i_max_corr++;
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
- isf[i - 2 - i_max_corr];
/* Calculate an estimate for ISF(18) and scale ISF based on the error */
est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
(isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
diff_isf[j] = scale * (isf[i] - isf[i - 1]);
/* Stability insurance */
for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
if (diff_isf[i] > diff_isf[i - 1]) {
diff_isf[i - 1] = 5.0 - diff_isf[i];
} else
diff_isf[i] = 5.0 - diff_isf[i - 1];
}
for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
/* Scale the ISF vector for 16000 Hz */
for (i = 0; i < LP_ORDER_16k - 1; i++)
isf[i] *= 0.8;
}
/**
* Spectral expand the LP coefficients using the equation:
* y[i] = x[i] * (gamma ** i)
*
* @param[out] out Output buffer (may use input array)
* @param[in] lpc LP coefficients array
* @param[in] gamma Weighting factor
* @param[in] size LP array size
*/
static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
{
int i;
float fac = gamma;
for (i = 0; i < size; i++) {
out[i] = lpc[i] * fac;
fac *= gamma;
}
}
/**
* Conduct 20th order linear predictive coding synthesis for the high
* frequency band excitation at 16kHz.
*
* @param[in] ctx The context
* @param[in] subframe Current subframe index (0 to 3)
* @param[in,out] samples Pointer to the output speech samples
* @param[in] exc Generated white-noise scaled excitation
* @param[in] isf Current frame isf vector
* @param[in] isf_past Past frame final isf vector
*/
static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
const float *exc, const float *isf, const float *isf_past)
{
float hb_lpc[LP_ORDER_16k];
enum Mode mode = ctx->fr_cur_mode;
if (mode == MODE_6k60) {
float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
double e_isp[LP_ORDER_16k];
ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1.0 - isfp_inter[subframe], LP_ORDER);
extrapolate_isf(e_isf);
e_isf[LP_ORDER_16k - 1] *= 2.0;
ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
} else {
lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
}
ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
(mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
}
/**
* Apply a 15th order filter to high-band samples.
* The filter characteristic depends on the given coefficients.
*
* @param[out] out Buffer for filtered output
* @param[in] fir_coef Filter coefficients
* @param[in,out] mem State from last filtering (updated)
* @param[in] in Input speech data (high-band)
*
* @remark It is safe to pass the same array in in and out parameters
*/
#ifndef hb_fir_filter
static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
float mem[HB_FIR_SIZE], const float *in)
{
int i, j;
float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
out[i] = 0.0;
for (j = 0; j <= HB_FIR_SIZE; j++)
out[i] += data[i + j] * fir_coef[j];
}
memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
}
#endif /* hb_fir_filter */
/**
* Update context state before the next subframe.
*/
static void update_sub_state(AMRWBContext *ctx)
{
memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
(AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
LP_ORDER * sizeof(float));
memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
UPS_MEM_SIZE * sizeof(float));
memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
LP_ORDER_16k * sizeof(float));
}
static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AMRWBContext *ctx = avctx->priv_data;
AVFrame *frame = data;
AMRWBFrame *cf = &ctx->frame;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int expected_fr_size, header_size;
float *buf_out;
float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
float fixed_gain_factor; // fixed gain correction factor (gamma)
float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
float synth_fixed_gain; // the fixed gain that synthesis should use
float voice_fac, stab_fac; // parameters used for gain smoothing
float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
float hb_gain;
int sub, i, ret;
/* get output buffer */
frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
buf_out = (float *)frame->data[0];
header_size = decode_mime_header(ctx, buf);
expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
if (!ctx->fr_quality)
av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
if (ctx->fr_cur_mode == NO_DATA || !ctx->fr_quality) {
/* The specification suggests a "random signal" and
"a muting technique" to "gradually decrease the output level". */
av_samples_set_silence(&frame->data[0], 0, frame->nb_samples, 1, AV_SAMPLE_FMT_FLT);
*got_frame_ptr = 1;
return expected_fr_size;
}
if (ctx->fr_cur_mode > MODE_SID) {
av_log(avctx, AV_LOG_ERROR,
"Invalid mode %d\n", ctx->fr_cur_mode);
return AVERROR_INVALIDDATA;
}
if (buf_size < expected_fr_size) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
*got_frame_ptr = 0;
return AVERROR_INVALIDDATA;
}
if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
avpriv_request_sample(avctx, "SID mode");
return AVERROR_PATCHWELCOME;
}
ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
/* Decode the quantized ISF vector */
if (ctx->fr_cur_mode == MODE_6k60) {
decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
} else {
decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
}
isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
ctx->isf_cur[LP_ORDER - 1] *= 2.0;
ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
/* Generate a ISP vector for each subframe */
if (ctx->first_frame) {
ctx->first_frame = 0;
memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
}
interpolate_isp(ctx->isp, ctx->isp_sub4_past);
for (sub = 0; sub < 4; sub++)
ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
for (sub = 0; sub < 4; sub++) {
const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
/* Decode adaptive codebook (pitch vector) */
decode_pitch_vector(ctx, cur_subframe, sub);
/* Decode innovative codebook (fixed vector) */
decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
cur_subframe->pul_il, ctx->fr_cur_mode);
pitch_sharpening(ctx, ctx->fixed_vector);
decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
&fixed_gain_factor, &ctx->pitch_gain[0]);
ctx->fixed_gain[0] =
ff_amr_set_fixed_gain(fixed_gain_factor,
ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
ctx->fixed_vector,
AMRWB_SFR_SIZE) /
AMRWB_SFR_SIZE,
ctx->prediction_error,
ENERGY_MEAN, energy_pred_fac);
/* Calculate voice factor and store tilt for next subframe */
voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
ctx->fixed_vector, ctx->fixed_gain[0],
&ctx->celpm_ctx);
ctx->tilt_coef = voice_fac * 0.25 + 0.25;
/* Construct current excitation */
for (i = 0; i < AMRWB_SFR_SIZE; i++) {
ctx->excitation[i] *= ctx->pitch_gain[0];
ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
ctx->excitation[i] = truncf(ctx->excitation[i]);
}
/* Post-processing of excitation elements */
synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
voice_fac, stab_fac);
synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
spare_vector);
pitch_enhancer(synth_fixed_vector, voice_fac);
synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
/* Synthesis speech post-processing */
de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
&ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
&ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
/* High frequency band (6.4 - 7.0 kHz) generation part */
ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
&ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
hb_gain = find_hb_gain(ctx, hb_samples,
cur_subframe->hb_gain, cf->vad);
scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
hb_exc, ctx->isf_cur, ctx->isf_past_final);
/* High-band post-processing filters */
hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
&ctx->samples_hb[LP_ORDER_16k]);
if (ctx->fr_cur_mode == MODE_23k85)
hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
hb_samples);
/* Add the low and high frequency bands */
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
/* Update buffers and history */
update_sub_state(ctx);
}
/* update state for next frame */
memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
*got_frame_ptr = 1;
return expected_fr_size;
}
const AVCodec ff_amrwb_decoder = {
.name = "amrwb",
.long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AMR_WB,
.priv_data_size = sizeof(AMRWBContext),
.init = amrwb_decode_init,
.decode = amrwb_decode_frame,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};
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