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ffmpeg/libavcodec/amrnbdec.c

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/*
* AMR narrowband decoder
* Copyright (c) 2006-2007 Robert Swain
* Copyright (c) 2009 Colin McQuillan
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AMR narrowband decoder
*
* This decoder uses floats for simplicity and so is not bit-exact. One
* difference is that differences in phase can accumulate. The test sequences
* in 3GPP TS 26.074 can still be useful.
*
* - Comparing this file's output to the output of the ref decoder gives a
* PSNR of 30 to 80. Plotting the output samples shows a difference in
* phase in some areas.
*
* - Comparing both decoders against their input, this decoder gives a similar
* PSNR. If the test sequence homing frames are removed (this decoder does
* not detect them), the PSNR is at least as good as the reference on 140
* out of 169 tests.
*/
#include <string.h>
#include <math.h>
#include "avcodec.h"
#include "libavutil/common.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "lsp.h"
#include "amr.h"
#include "amrnbdata.h"
#define AMR_BLOCK_SIZE 160 ///< samples per frame
#define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow
/**
* Scale from constructed speech to [-1,1]
*
* AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
* upscales by two (section 6.2.2).
*
* Fundamentally, this scale is determined by energy_mean through
* the fixed vector contribution to the excitation vector.
*/
#define AMR_SAMPLE_SCALE (2.0 / 32768.0)
/** Prediction factor for 12.2kbit/s mode */
#define PRED_FAC_MODE_12k2 0.65
#define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz
#define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter
#define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode
/** Initial energy in dB. Also used for bad frames (unimplemented). */
#define MIN_ENERGY -14.0
/** Maximum sharpening factor
*
* The specification says 0.8, which should be 13107, but the reference C code
* uses 13017 instead. (Amusingly the same applies to SHARP_MAX in bitexact G.729.)
*/
#define SHARP_MAX 0.79449462890625
/** Number of impulse response coefficients used for tilt factor */
#define AMR_TILT_RESPONSE 22
/** Tilt factor = 1st reflection coefficient * gamma_t */
#define AMR_TILT_GAMMA_T 0.8
/** Adaptive gain control factor used in post-filter */
#define AMR_AGC_ALPHA 0.9
typedef struct AMRContext {
AVFrame avframe; ///< AVFrame for decoded samples
AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
enum Mode cur_frame_mode;
int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
float *excitation; ///< pointer to the current excitation vector in excitation_buf
float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
uint8_t hang_count; ///< the number of subframes since a hangover period started
float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
uint8_t ir_filter_onset; ///< flag for impulse response filter strength
float postfilter_mem[10]; ///< previous intermediate values in the formant filter
float tilt_mem; ///< previous input to tilt compensation filter
float postfilter_agc; ///< previous factor used for adaptive gain control
float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
} AMRContext;
/** Double version of ff_weighted_vector_sumf() */
static void weighted_vector_sumd(double *out, const double *in_a,
const double *in_b, double weight_coeff_a,
double weight_coeff_b, int length)
{
int i;
for (i = 0; i < length; i++)
out[i] = weight_coeff_a * in_a[i]
+ weight_coeff_b * in_b[i];
}
static av_cold int amrnb_decode_init(AVCodecContext *avctx)
{
AMRContext *p = avctx->priv_data;
int i;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
// p->excitation always points to the same position in p->excitation_buf
p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
for (i = 0; i < LP_FILTER_ORDER; i++) {
p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
}
for (i = 0; i < 4; i++)
p->prediction_error[i] = MIN_ENERGY;
avcodec_get_frame_defaults(&p->avframe);
avctx->coded_frame = &p->avframe;
return 0;
}
/**
* Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
*
* The order of speech bits is specified by 3GPP TS 26.101.
*
* @param p the context
* @param buf pointer to the input buffer
* @param buf_size size of the input buffer
*
* @return the frame mode
*/
static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
int buf_size)
{
enum Mode mode;
// Decode the first octet.
mode = buf[0] >> 3 & 0x0F; // frame type
p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit
if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
return NO_DATA;
}
if (mode < MODE_DTX)
ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
amr_unpacking_bitmaps_per_mode[mode]);
return mode;
}
/// @name AMR pitch LPC coefficient decoding functions
/// @{
/**
* Interpolate the LSF vector (used for fixed gain smoothing).
* The interpolation is done over all four subframes even in MODE_12k2.
*
* @param[in,out] lsf_q LSFs in [0,1] for each subframe
* @param[in] lsf_new New LSFs in [0,1] for subframe 4
*/
static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
{
int i;
for (i = 0; i < 4; i++)
ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
0.25 * (3 - i), 0.25 * (i + 1),
LP_FILTER_ORDER);
}
/**
* Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
*
* @param p the context
* @param lsp output LSP vector
* @param lsf_no_r LSF vector without the residual vector added
* @param lsf_quantizer pointers to LSF dictionary tables
* @param quantizer_offset offset in tables
* @param sign for the 3 dictionary table
* @param update store data for computing the next frame's LSFs
*/
static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
const float lsf_no_r[LP_FILTER_ORDER],
const int16_t *lsf_quantizer[5],
const int quantizer_offset,
const int sign, const int update)
{
int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
int i;
for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
2 * sizeof(*lsf_r));
if (sign) {
lsf_r[4] *= -1;
lsf_r[5] *= -1;
}
if (update)
memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
for (i = 0; i < LP_FILTER_ORDER; i++)
lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
if (update)
interpolate_lsf(p->lsf_q, lsf_q);
ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
}
/**
* Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
*
* @param p pointer to the AMRContext
*/
static void lsf2lsp_5(AMRContext *p)
{
const uint16_t *lsf_param = p->frame.lsf;
float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
const int16_t *lsf_quantizer[5];
int i;
lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
for (i = 0; i < LP_FILTER_ORDER; i++)
lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
// interpolate LSP vectors at subframes 1 and 3
weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
}
/**
* Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
*
* @param p pointer to the AMRContext
*/
static void lsf2lsp_3(AMRContext *p)
{
const uint16_t *lsf_param = p->frame.lsf;
int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
const int16_t *lsf_quantizer;
int i, j;
lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
// calculate mean-removed LSF vector and add mean
for (i = 0; i < LP_FILTER_ORDER; i++)
lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);
ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
// store data for computing the next frame's LSFs
interpolate_lsf(p->lsf_q, lsf_q);
memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
// interpolate LSP vectors at subframes 1, 2 and 3
for (i = 1; i <= 3; i++)
for(j = 0; j < LP_FILTER_ORDER; j++)
p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
(p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
}
/// @}
/// @name AMR pitch vector decoding functions
/// @{
/**
* Like ff_decode_pitch_lag(), but with 1/6 resolution
*/
static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
const int prev_lag_int, const int subframe)
{
if (subframe == 0 || subframe == 2) {
if (pitch_index < 463) {
*lag_int = (pitch_index + 107) * 10923 >> 16;
*lag_frac = pitch_index - *lag_int * 6 + 105;
} else {
*lag_int = pitch_index - 368;
*lag_frac = 0;
}
} else {
*lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
*lag_frac = pitch_index - *lag_int * 6 - 3;
*lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
PITCH_DELAY_MAX - 9);
}
}
static void decode_pitch_vector(AMRContext *p,
const AMRNBSubframe *amr_subframe,
const int subframe)
{
int pitch_lag_int, pitch_lag_frac;
enum Mode mode = p->cur_frame_mode;
if (p->cur_frame_mode == MODE_12k2) {
decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
amr_subframe->p_lag, p->pitch_lag_int,
subframe);
} else
ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
amr_subframe->p_lag,
p->pitch_lag_int, subframe,
mode != MODE_4k75 && mode != MODE_5k15,
mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
pitch_lag_int += pitch_lag_frac > 0;
/* Calculate the pitch vector by interpolating the past excitation at the
pitch lag using a b60 hamming windowed sinc function. */
ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
ff_b60_sinc, 6,
pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
10, AMR_SUBFRAME_SIZE);
memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
}
/// @}
/// @name AMR algebraic code book (fixed) vector decoding functions
/// @{
/**
* Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
*/
static void decode_10bit_pulse(int code, int pulse_position[8],
int i1, int i2, int i3)
{
// coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
// the 3 pulses and the upper 7 bits being coded in base 5
const uint8_t *positions = base_five_table[code >> 3];
pulse_position[i1] = (positions[2] << 1) + ( code & 1);
pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
}
/**
* Decode the algebraic codebook index to pulse positions and signs and
* construct the algebraic codebook vector for MODE_10k2.
*
* @param fixed_index positions of the eight pulses
* @param fixed_sparse pointer to the algebraic codebook vector
*/
static void decode_8_pulses_31bits(const int16_t *fixed_index,
AMRFixed *fixed_sparse)
{
int pulse_position[8];
int i, temp;
decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
// coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
// the 2 pulses and the upper 5 bits being coded in base 5
temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
pulse_position[3] = temp % 5;
pulse_position[7] = temp / 5;
if (pulse_position[7] & 1)
pulse_position[3] = 4 - pulse_position[3];
pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
fixed_sparse->n = 8;
for (i = 0; i < 4; i++) {
const int pos1 = (pulse_position[i] << 2) + i;
const int pos2 = (pulse_position[i + 4] << 2) + i;
const float sign = fixed_index[i] ? -1.0 : 1.0;
fixed_sparse->x[i ] = pos1;
fixed_sparse->x[i + 4] = pos2;
fixed_sparse->y[i ] = sign;
fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
}
}
/**
* Decode the algebraic codebook index to pulse positions and signs,
* then construct the algebraic codebook vector.
*
* nb of pulses | bits encoding pulses
* For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
* MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
* MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
* MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
*
* @param fixed_sparse pointer to the algebraic codebook vector
* @param pulses algebraic codebook indexes
* @param mode mode of the current frame
* @param subframe current subframe number
*/
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
const enum Mode mode, const int subframe)
{
assert(MODE_4k75 <= mode && mode <= MODE_12k2);
if (mode == MODE_12k2) {
ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
} else if (mode == MODE_10k2) {
decode_8_pulses_31bits(pulses, fixed_sparse);
} else {
int *pulse_position = fixed_sparse->x;
int i, pulse_subset;
const int fixed_index = pulses[0];
if (mode <= MODE_5k15) {
pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
fixed_sparse->n = 2;
} else if (mode == MODE_5k9) {
pulse_subset = ((fixed_index & 1) << 1) + 1;
pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
pulse_subset = (fixed_index >> 4) & 3;
pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
} else if (mode == MODE_6k7) {
pulse_position[0] = (fixed_index & 7) * 5;
pulse_subset = (fixed_index >> 2) & 2;
pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
pulse_subset = (fixed_index >> 6) & 2;
pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
fixed_sparse->n = 3;
} else { // mode <= MODE_7k95
pulse_position[0] = gray_decode[ fixed_index & 7];
pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
pulse_subset = (fixed_index >> 9) & 1;
pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
fixed_sparse->n = 4;
}
for (i = 0; i < fixed_sparse->n; i++)
fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
}
}
/**
* Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
*
* @param p the context
* @param subframe unpacked amr subframe
* @param mode mode of the current frame
* @param fixed_sparse sparse respresentation of the fixed vector
*/
static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
AMRFixed *fixed_sparse)
{
// The spec suggests the current pitch gain is always used, but in other
// modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
// so the codebook gain cannot depend on the quantized pitch gain.
if (mode == MODE_12k2)
p->beta = FFMIN(p->pitch_gain[4], 1.0);
fixed_sparse->pitch_lag = p->pitch_lag_int;
fixed_sparse->pitch_fac = p->beta;
// Save pitch sharpening factor for the next subframe
// MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
// the fact that the gains for two subframes are jointly quantized.
if (mode != MODE_4k75 || subframe & 1)
p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
}
/// @}
/// @name AMR gain decoding functions
/// @{
/**
* fixed gain smoothing
* Note that where the spec specifies the "spectrum in the q domain"
* in section 6.1.4, in fact frequencies should be used.
*
* @param p the context
* @param lsf LSFs for the current subframe, in the range [0,1]
* @param lsf_avg averaged LSFs
* @param mode mode of the current frame
*
* @return fixed gain smoothed
*/
static float fixed_gain_smooth(AMRContext *p , const float *lsf,
const float *lsf_avg, const enum Mode mode)
{
float diff = 0.0;
int i;
for (i = 0; i < LP_FILTER_ORDER; i++)
diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
// If diff is large for ten subframes, disable smoothing for a 40-subframe
// hangover period.
p->diff_count++;
if (diff <= 0.65)
p->diff_count = 0;
if (p->diff_count > 10) {
p->hang_count = 0;
p->diff_count--; // don't let diff_count overflow
}
if (p->hang_count < 40) {
p->hang_count++;
} else if (mode < MODE_7k4 || mode == MODE_10k2) {
const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
p->fixed_gain[2] + p->fixed_gain[3] +
p->fixed_gain[4]) * 0.2;
return smoothing_factor * p->fixed_gain[4] +
(1.0 - smoothing_factor) * fixed_gain_mean;
}
return p->fixed_gain[4];
}
/**
* Decode pitch gain and fixed gain factor (part of section 6.1.3).
*
* @param p the context
* @param amr_subframe unpacked amr subframe
* @param mode mode of the current frame
* @param subframe current subframe number
* @param fixed_gain_factor decoded gain correction factor
*/
static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
const enum Mode mode, const int subframe,
float *fixed_gain_factor)
{
if (mode == MODE_12k2 || mode == MODE_7k95) {
p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
* (1.0 / 16384.0);
*fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
* (1.0 / 2048.0);
} else {
const uint16_t *gains;
if (mode >= MODE_6k7) {
gains = gains_high[amr_subframe->p_gain];
} else if (mode >= MODE_5k15) {
gains = gains_low [amr_subframe->p_gain];
} else {
// gain index is only coded in subframes 0,2 for MODE_4k75
gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
}
p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
*fixed_gain_factor = gains[1] * (1.0 / 4096.0);
}
}
/// @}
/// @name AMR preprocessing functions
/// @{
/**
* Circularly convolve a sparse fixed vector with a phase dispersion impulse
* response filter (D.6.2 of G.729 and 6.1.5 of AMR).
*
* @param out vector with filter applied
* @param in source vector
* @param filter phase filter coefficients
*
* out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
*/
static void apply_ir_filter(float *out, const AMRFixed *in,
const float *filter)
{
float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2
filter2[AMR_SUBFRAME_SIZE];
int lag = in->pitch_lag;
float fac = in->pitch_fac;
int i;
if (lag < AMR_SUBFRAME_SIZE) {
ff_celp_circ_addf(filter1, filter, filter, lag, fac,
AMR_SUBFRAME_SIZE);
if (lag < AMR_SUBFRAME_SIZE >> 1)
ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
AMR_SUBFRAME_SIZE);
}
memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
for (i = 0; i < in->n; i++) {
int x = in->x[i];
float y = in->y[i];
const float *filterp;
if (x >= AMR_SUBFRAME_SIZE - lag) {
filterp = filter;
} else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
filterp = filter1;
} else
filterp = filter2;
ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
}
}
/**
* Reduce fixed vector sparseness by smoothing with one of three IR filters.
* Also know as "adaptive phase dispersion".
*
* This implements 3GPP TS 26.090 section 6.1(5).
*
* @param p the context
* @param fixed_sparse algebraic codebook vector
* @param fixed_vector unfiltered fixed vector
* @param fixed_gain smoothed gain
* @param out space for modified vector if necessary
*/
static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
const float *fixed_vector,
float fixed_gain, float *out)
{
int ir_filter_nr;
if (p->pitch_gain[4] < 0.6) {
ir_filter_nr = 0; // strong filtering
} else if (p->pitch_gain[4] < 0.9) {
ir_filter_nr = 1; // medium filtering
} else
ir_filter_nr = 2; // no filtering
// detect 'onset'
if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
p->ir_filter_onset = 2;
} else if (p->ir_filter_onset)
p->ir_filter_onset--;
if (!p->ir_filter_onset) {
int i, count = 0;
for (i = 0; i < 5; i++)
if (p->pitch_gain[i] < 0.6)
count++;
if (count > 2)
ir_filter_nr = 0;
if (ir_filter_nr > p->prev_ir_filter_nr + 1)
ir_filter_nr--;
} else if (ir_filter_nr < 2)
ir_filter_nr++;
// Disable filtering for very low level of fixed_gain.
// Note this step is not specified in the technical description but is in
// the reference source in the function Ph_disp.
if (fixed_gain < 5.0)
ir_filter_nr = 2;
if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
&& ir_filter_nr < 2) {
apply_ir_filter(out, fixed_sparse,
(p->cur_frame_mode == MODE_7k95 ?
ir_filters_lookup_MODE_7k95 :
ir_filters_lookup)[ir_filter_nr]);
fixed_vector = out;
}
// update ir filter strength history
p->prev_ir_filter_nr = ir_filter_nr;
p->prev_sparse_fixed_gain = fixed_gain;
return fixed_vector;
}
/// @}
/// @name AMR synthesis functions
/// @{
/**
* Conduct 10th order linear predictive coding synthesis.
*
* @param p pointer to the AMRContext
* @param lpc pointer to the LPC coefficients
* @param fixed_gain fixed codebook gain for synthesis
* @param fixed_vector algebraic codebook vector
* @param samples pointer to the output speech samples
* @param overflow 16-bit overflow flag
*/
static int synthesis(AMRContext *p, float *lpc,
float fixed_gain, const float *fixed_vector,
float *samples, uint8_t overflow)
{
int i;
float excitation[AMR_SUBFRAME_SIZE];
// if an overflow has been detected, the pitch vector is scaled down by a
// factor of 4
if (overflow)
for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
p->pitch_vector[i] *= 0.25;
ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
// emphasize pitch vector contribution
if (p->pitch_gain[4] > 0.5 && !overflow) {
float energy = ff_dot_productf(excitation, excitation,
AMR_SUBFRAME_SIZE);
float pitch_factor =
p->pitch_gain[4] *
(p->cur_frame_mode == MODE_12k2 ?
0.25 * FFMIN(p->pitch_gain[4], 1.0) :
0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
excitation[i] += pitch_factor * p->pitch_vector[i];
ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
AMR_SUBFRAME_SIZE);
}
ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
LP_FILTER_ORDER);
// detect overflow
for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
return 1;
}
return 0;
}
/// @}
/// @name AMR update functions
/// @{
/**
* Update buffers and history at the end of decoding a subframe.
*
* @param p pointer to the AMRContext
*/
static void update_state(AMRContext *p)
{
memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
(PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
LP_FILTER_ORDER * sizeof(float));
}
/// @}
/// @name AMR Postprocessing functions
/// @{
/**
* Get the tilt factor of a formant filter from its transfer function
*
* @param lpc_n LP_FILTER_ORDER coefficients of the numerator
* @param lpc_d LP_FILTER_ORDER coefficients of the denominator
*/
static float tilt_factor(float *lpc_n, float *lpc_d)
{
float rh0, rh1; // autocorrelation at lag 0 and 1
// LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
hf[0] = 1.0;
memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
LP_FILTER_ORDER);
rh0 = ff_dot_productf(hf, hf, AMR_TILT_RESPONSE);
rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
// The spec only specifies this check for 12.2 and 10.2 kbit/s
// modes. But in the ref source the tilt is always non-negative.
return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
}
/**
* Perform adaptive post-filtering to enhance the quality of the speech.
* See section 6.2.1.
*
* @param p pointer to the AMRContext
* @param lpc interpolated LP coefficients for this subframe
* @param buf_out output of the filter
*/
static void postfilter(AMRContext *p, float *lpc, float *buf_out)
{
int i;
float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
float speech_gain = ff_dot_productf(samples, samples,
AMR_SUBFRAME_SIZE);
float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
const float *gamma_n, *gamma_d; // Formant filter factor table
float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
gamma_n = ff_pow_0_7;
gamma_d = ff_pow_0_75;
} else {
gamma_n = ff_pow_0_55;
gamma_d = ff_pow_0_7;
}
for (i = 0; i < LP_FILTER_ORDER; i++) {
lpc_n[i] = lpc[i] * gamma_n[i];
lpc_d[i] = lpc[i] * gamma_d[i];
}
memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
sizeof(float) * LP_FILTER_ORDER);
ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
pole_out + LP_FILTER_ORDER,
AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
AMR_SUBFRAME_SIZE);
ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
AMR_AGC_ALPHA, &p->postfilter_agc);
}
/// @}
static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AMRContext *p = avctx->priv_data; // pointer to private data
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
float *buf_out; // pointer to the output data buffer
int i, subframe, ret;
float fixed_gain_factor;
AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
float synth_fixed_gain; // the fixed gain that synthesis should use
const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
/* get output buffer */
p->avframe.nb_samples = AMR_BLOCK_SIZE;
if ((ret = avctx->get_buffer(avctx, &p->avframe)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
buf_out = (float *)p->avframe.data[0];
p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
if (p->cur_frame_mode == NO_DATA) {
av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
return AVERROR_INVALIDDATA;
}
if (p->cur_frame_mode == MODE_DTX) {
av_log_missing_feature(avctx, "dtx mode", 1);
return -1;
}
if (p->cur_frame_mode == MODE_12k2) {
lsf2lsp_5(p);
} else
lsf2lsp_3(p);
for (i = 0; i < 4; i++)
ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
for (subframe = 0; subframe < 4; subframe++) {
const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
decode_pitch_vector(p, amr_subframe, subframe);
decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
p->cur_frame_mode, subframe);
// The fixed gain (section 6.1.3) depends on the fixed vector
// (section 6.1.2), but the fixed vector calculation uses
// pitch sharpening based on the on the pitch gain (section 6.1.3).
// So the correct order is: pitch gain, pitch sharpening, fixed gain.
decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
&fixed_gain_factor);
pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
if (fixed_sparse.pitch_lag == 0) {
av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
return AVERROR_INVALIDDATA;
}
ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
AMR_SUBFRAME_SIZE);
p->fixed_gain[4] =
ff_amr_set_fixed_gain(fixed_gain_factor,
ff_dot_productf(p->fixed_vector, p->fixed_vector,
AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
p->prediction_error,
energy_mean[p->cur_frame_mode], energy_pred_fac);
// The excitation feedback is calculated without any processing such
// as fixed gain smoothing. This isn't mentioned in the specification.
for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
p->excitation[i] *= p->pitch_gain[4];
ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
AMR_SUBFRAME_SIZE);
// In the ref decoder, excitation is stored with no fractional bits.
// This step prevents buzz in silent periods. The ref encoder can
// emit long sequences with pitch factor greater than one. This
// creates unwanted feedback if the excitation vector is nonzero.
// (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
p->excitation[i] = truncf(p->excitation[i]);
// Smooth fixed gain.
// The specification is ambiguous, but in the reference source, the
// smoothed value is NOT fed back into later fixed gain smoothing.
synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
p->lsf_avg, p->cur_frame_mode);
synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
synth_fixed_gain, spare_vector);
if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
// overflow detected -> rerun synthesis scaling pitch vector down
// by a factor of 4, skipping pitch vector contribution emphasis
// and adaptive gain control
synthesis(p, p->lpc[subframe], synth_fixed_gain,
synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
// update buffers and history
ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
update_state(p);
}
ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
highpass_poles,
highpass_gain * AMR_SAMPLE_SCALE,
p->high_pass_mem, AMR_BLOCK_SIZE);
/* Update averaged lsf vector (used for fixed gain smoothing).
*
* Note that lsf_avg should not incorporate the current frame's LSFs
* for fixed_gain_smooth.
* The specification has an incorrect formula: the reference decoder uses
* qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
0.84, 0.16, LP_FILTER_ORDER);
*got_frame_ptr = 1;
*(AVFrame *)data = p->avframe;
/* return the amount of bytes consumed if everything was OK */
return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
}
AVCodec ff_amrnb_decoder = {
.name = "amrnb",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AMR_NB,
.priv_data_size = sizeof(AMRContext),
.init = amrnb_decode_init,
.decode = amrnb_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
};
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