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FFmpeg/libavcodec/amrwbdec.c

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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) << 1;
/* 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);
}
/**
* 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;
if (ctx->fr_cur_mode == MODE_23k85)
return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
/* return gain bounded by [0.1, 1.0] */
return av_clipf((1.0 - FFMAX(0.0, 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);
if (ctx->fr_cur_mode > MODE_SID) {
av_log(avctx, AV_LOG_ERROR,
"Invalid mode %d\n", ctx->fr_cur_mode);
return AVERROR_INVALIDDATA;
}
expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
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_quality || ctx->fr_cur_mode > MODE_SID)
av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
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;
}
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,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};