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ffmpeg/libavcodec/amrnbdec.c
Nedeljko Babic 3827a86eac Optimization of AMR NB and WB decoders for MIPS 
AMR NB and WB decoders are optimized for MIPS architecture.
Appropriate Makefiles are changed accordingly.

Cnfigure script is changed in order to support optimizations.
 Optimizations are enabled by default when compiling is done for
  mips architecture.
 Appropriate cflags are automatically set.
 Support for several mips CPUs is added in configure script.

New ffmpeg options are added for disabling optimizations.

The FFMPEG option --disable-mipsfpu disables MIPS floating point
 optimizations.
The FFMPEG option --disable-mips32r2 disables MIPS32R2
 optimizations.
The FFMPEG option --disable-mipsdspr1 disables MIPS DSP ASE R1
 optimizations.
The FFMPEG option --disable-mipsdspr2 disables MIPS DSP ASE R2
 optimizations.

Signed-off-by: Nedeljko Babic <nbabic@mips.com>
Reviewed-by: Vitor Sessak <vitor1001@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-11 21:12:39 +02:00

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/*
* AMR narrowband decoder
* Copyright (c) 2006-2007 Robert Swain
* Copyright (c) 2009 Colin McQuillan
*
* 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 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 "libavutil/avassert.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 g729dec.c.)
*/
#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
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
} 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;
ff_acelp_filter_init(&p->acelpf_ctx);
ff_acelp_vectors_init(&p->acelpv_ctx);
ff_celp_filter_init(&p->celpf_ctx);
ff_celp_math_init(&p->celpm_ctx);
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] ctx The Context
* @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(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
{
int i;
for (i = 0; i < 4; i++)
ctx->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->acelpv_ctx, 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->acelpv_ctx, 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. */
p->acelpf_ctx.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)
{
av_assert1(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;
p->acelpv_ctx.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 = p->celpm_ctx.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);
}
p->celpf_ctx.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 p The Context
* @param lpc_n LP_FILTER_ORDER coefficients of the numerator
* @param lpc_d LP_FILTER_ORDER coefficients of the denominator
*/
static float tilt_factor(AMRContext *p, 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);
p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
AMR_TILT_RESPONSE,
LP_FILTER_ORDER);
rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE);
rh1 = p->celpm_ctx.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 = p->celpm_ctx.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);
p->celpf_ctx.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);
p->celpf_ctx.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(p, 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", 0);
av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
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,
p->celpm_ctx.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);
}
p->acelpf_ctx.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. */
p->acelpv_ctx.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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