libavcodec/amrnbdec.c File Reference

AMR narrowband decoder. More...

#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"

Go to the source code of this file.

Data Structures

struct  AMRContext

Defines

#define AMR_BLOCK_SIZE   160
 samples per frame
#define AMR_SAMPLE_BOUND   32768.0
 threshold for synthesis overflow
#define AMR_SAMPLE_SCALE   (2.0 / 32768.0)
 Scale from constructed speech to [-1,1].
#define PRED_FAC_MODE_12k2   0.65
 Prediction factor for 12.2kbit/s mode.
#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.
#define MIN_ENERGY   -14.0
 Initial energy in dB.
#define SHARP_MAX   0.79449462890625
 Maximum sharpening factor.
#define AMR_TILT_RESPONSE   22
 Number of impulse response coefficients used for tilt factor.
#define AMR_TILT_GAMMA_T   0.8
 Tilt factor = 1st reflection coefficient * gamma_t.
#define AMR_AGC_ALPHA   0.9
 Adaptive gain control factor used in post-filter.

Functions

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)
 Double version of ff_weighted_vector_sumf().
static av_cold int amrnb_decode_init (AVCodecContext *avctx)
static enum Mode unpack_bitstream (AMRContext *p, const uint8_t *buf, int buf_size)
 Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
static int amrnb_decode_frame (AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
AMR pitch LPC coefficient decoding functions
static void interpolate_lsf (float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
 Interpolate the LSF vector (used for fixed gain smoothing).
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)
 Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
static void lsf2lsp_5 (AMRContext *p)
 Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
static void lsf2lsp_3 (AMRContext *p)
 Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
AMR pitch vector decoding functions
static void decode_pitch_lag_1_6 (int *lag_int, int *lag_frac, int pitch_index, const int prev_lag_int, const int subframe)
 Like ff_decode_pitch_lag(), but with 1/6 resolution.
static void decode_pitch_vector (AMRContext *p, const AMRNBSubframe *amr_subframe, const int subframe)
AMR algebraic code book (fixed) vector decoding functions
static void decode_10bit_pulse (int code, int pulse_position[8], int i1, int i2, int i3)
 Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
static void decode_8_pulses_31bits (const int16_t *fixed_index, AMRFixed *fixed_sparse)
 Decode the algebraic codebook index to pulse positions and signs and construct the algebraic codebook vector for MODE_10k2.
static void decode_fixed_sparse (AMRFixed *fixed_sparse, const uint16_t *pulses, const enum Mode mode, const int subframe)
 Decode the algebraic codebook index to pulse positions and signs, then construct the algebraic codebook vector.
static void pitch_sharpening (AMRContext *p, int subframe, enum Mode mode, AMRFixed *fixed_sparse)
 Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2).
AMR gain decoding functions
static float fixed_gain_smooth (AMRContext *p, const float *lsf, const float *lsf_avg, const enum Mode mode)
 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.
static void decode_gains (AMRContext *p, const AMRNBSubframe *amr_subframe, const enum Mode mode, const int subframe, float *fixed_gain_factor)
 Decode pitch gain and fixed gain factor (part of section 6.1.3).
AMR preprocessing functions
static void apply_ir_filter (float *out, const AMRFixed *in, const float *filter)
 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).
static const float * anti_sparseness (AMRContext *p, AMRFixed *fixed_sparse, const float *fixed_vector, float fixed_gain, float *out)
 Reduce fixed vector sparseness by smoothing with one of three IR filters.
AMR synthesis functions
static int synthesis (AMRContext *p, float *lpc, float fixed_gain, const float *fixed_vector, float *samples, uint8_t overflow)
 Conduct 10th order linear predictive coding synthesis.
AMR update functions
static void update_state (AMRContext *p)
 Update buffers and history at the end of decoding a subframe.
AMR Postprocessing functions
static float tilt_factor (float *lpc_n, float *lpc_d)
 Get the tilt factor of a formant filter from its transfer function.
static void postfilter (AMRContext *p, float *lpc, float *buf_out)
 Perform adaptive post-filtering to enhance the quality of the speech.

Variables

AVCodec ff_amrnb_decoder


Detailed Description

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.

Definition in file amrnbdec.c.


Define Documentation

#define AMR_AGC_ALPHA   0.9

Adaptive gain control factor used in post-filter.

Definition at line 94 of file amrnbdec.c.

Referenced by postfilter().

#define AMR_BLOCK_SIZE   160

samples per frame

Definition at line 58 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

#define AMR_SAMPLE_BOUND   32768.0

threshold for synthesis overflow

Definition at line 59 of file amrnbdec.c.

Referenced by synthesis().

#define AMR_SAMPLE_SCALE   (2.0 / 32768.0)

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.

Definition at line 70 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

#define AMR_TILT_GAMMA_T   0.8

Tilt factor = 1st reflection coefficient * gamma_t.

Definition at line 92 of file amrnbdec.c.

Referenced by tilt_factor().

#define AMR_TILT_RESPONSE   22

Number of impulse response coefficients used for tilt factor.

Definition at line 90 of file amrnbdec.c.

Referenced by tilt_factor().

#define LSF_R_FAC   (8000.0 / 32768.0)

LSF residual tables to Hertz.

Definition at line 75 of file amrnbdec.c.

Referenced by lsf2lsp_3(), lsf2lsp_5(), and lsf2lsp_for_mode12k2().

#define MIN_ENERGY   -14.0

Initial energy in dB.

Also used for bad frames (unimplemented).

Definition at line 80 of file amrnbdec.c.

Referenced by amrnb_decode_init(), and amrwb_decode_init().

#define MIN_LSF_SPACING   (50.0488 / 8000.0)

Ensures stability of LPC filter.

Definition at line 76 of file amrnbdec.c.

Referenced by lsf2lsp_3(), and lsf2lsp_for_mode12k2().

#define PITCH_LAG_MIN_MODE_12k2   18

Lower bound on decoded lag search in 12.2kbit/s mode.

Definition at line 77 of file amrnbdec.c.

Referenced by decode_pitch_lag_1_6().

#define PRED_FAC_MODE_12k2   0.65

Prediction factor for 12.2kbit/s mode.

Definition at line 73 of file amrnbdec.c.

Referenced by lsf2lsp_5().

#define SHARP_MAX   0.79449462890625

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.)

Definition at line 87 of file amrnbdec.c.

Referenced by decode_frame(), pitch_sharpening(), and synthesis().


Function Documentation

static int amrnb_decode_frame ( AVCodecContext avctx,
void *  data,
int *  got_frame_ptr,
AVPacket avpkt 
) [static]

Definition at line 924 of file amrnbdec.c.

static av_cold int amrnb_decode_init ( AVCodecContext avctx  )  [static]

Definition at line 152 of file amrnbdec.c.

static const float* anti_sparseness ( AMRContext p,
AMRFixed fixed_sparse,
const float *  fixed_vector,
float  fixed_gain,
float *  out 
) [static]

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).

Parameters:
p the context
fixed_sparse algebraic codebook vector
fixed_vector unfiltered fixed vector
fixed_gain smoothed gain
out space for modified vector if necessary

Definition at line 698 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), and amrwb_decode_frame().

static void apply_ir_filter ( float *  out,
const AMRFixed in,
const float *  filter 
) [static]

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).

Parameters:
out vector with filter applied
in source vector
filter phase filter coefficients
out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)len] }

< filters at pitch lag*1 and *2

Definition at line 651 of file amrnbdec.c.

Referenced by anti_sparseness().

static void decode_10bit_pulse ( int  code,
int  pulse_position[8],
int  i1,
int  i2,
int  i3 
) [static]

Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.

Definition at line 414 of file amrnbdec.c.

Referenced by decode_8_pulses_31bits().

static void decode_8_pulses_31bits ( const int16_t fixed_index,
AMRFixed fixed_sparse 
) [static]

Decode the algebraic codebook index to pulse positions and signs and construct the algebraic codebook vector for MODE_10k2.

Parameters:
fixed_index positions of the eight pulses
fixed_sparse pointer to the algebraic codebook vector

Definition at line 432 of file amrnbdec.c.

Referenced by decode_fixed_sparse().

static void decode_fixed_sparse ( AMRFixed fixed_sparse,
const uint16_t *  pulses,
const enum Mode  mode,
const int  subframe 
) [static]

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

Parameters:
fixed_sparse pointer to the algebraic codebook vector
pulses algebraic codebook indexes
mode mode of the current frame
subframe current subframe number

Definition at line 478 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), and decode_frame().

static void decode_gains ( AMRContext p,
const AMRNBSubframe amr_subframe,
const enum Mode  mode,
const int  subframe,
float *  fixed_gain_factor 
) [static]

Decode pitch gain and fixed gain factor (part of section 6.1.3).

Parameters:
p the context
amr_subframe unpacked amr subframe
mode mode of the current frame
subframe current subframe number
fixed_gain_factor decoded gain correction factor

Definition at line 609 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), and amrwb_decode_frame().

static void decode_pitch_lag_1_6 ( int *  lag_int,
int *  lag_frac,
int  pitch_index,
const int  prev_lag_int,
const int  subframe 
) [static]

Like ff_decode_pitch_lag(), but with 1/6 resolution.

Definition at line 352 of file amrnbdec.c.

Referenced by decode_pitch_vector().

static void decode_pitch_vector ( AMRContext p,
const AMRNBSubframe amr_subframe,
const int  subframe 
) [static]

Definition at line 371 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), and amrwb_decode_frame().

static float fixed_gain_smooth ( AMRContext p,
const float *  lsf,
const float *  lsf_avg,
const enum Mode  mode 
) [static]

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.

Parameters:
p the context
lsf LSFs for the current subframe, in the range [0,1]
lsf_avg averaged LSFs
mode mode of the current frame
Returns:
fixed gain smoothed

Definition at line 567 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

static void interpolate_lsf ( float  lsf_q[4][LP_FILTER_ORDER],
float *  lsf_new 
) [static]

Interpolate the LSF vector (used for fixed gain smoothing).

The interpolation is done over all four subframes even in MODE_12k2.

Parameters:
[in,out] lsf_q LSFs in [0,1] for each subframe
[in] lsf_new New LSFs in [0,1] for subframe 4

Definition at line 219 of file amrnbdec.c.

Referenced by lsf2lsp_3(), and lsf2lsp_for_mode12k2().

static void lsf2lsp_3 ( AMRContext p  )  [static]

Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.

Parameters:
p pointer to the AMRContext

Definition at line 307 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

static void lsf2lsp_5 ( AMRContext p  )  [static]

Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.

Parameters:
p pointer to the AMRContext

Definition at line 278 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

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 
) [static]

Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.

Parameters:
p the context
lsp output LSP vector
lsf_no_r LSF vector without the residual vector added
lsf_quantizer pointers to LSF dictionary tables
quantizer_offset offset in tables
sign for the 3 dictionary table
update store data for computing the next frame's LSFs

Definition at line 240 of file amrnbdec.c.

Referenced by lsf2lsp_5().

static void pitch_sharpening ( AMRContext p,
int  subframe,
enum Mode  mode,
AMRFixed fixed_sparse 
) [static]

Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2).

Parameters:
p the context
subframe unpacked amr subframe
mode mode of the current frame
fixed_sparse sparse respresentation of the fixed vector

Definition at line 531 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), amrwb_decode_frame(), and eval_ir().

static void postfilter ( AMRContext p,
float *  lpc,
float *  buf_out 
) [static]

Perform adaptive post-filtering to enhance the quality of the speech.

See section 6.2.1.

Parameters:
p pointer to the AMRContext
lpc interpolated LP coefficients for this subframe
buf_out output of the filter

Definition at line 880 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), ff_sipr_decode_frame_16k(), qcelp_decode_frame(), and synth_frame().

static int synthesis ( AMRContext p,
float *  lpc,
float  fixed_gain,
const float *  fixed_vector,
float *  samples,
uint8_t  overflow 
) [static]

Conduct 10th order linear predictive coding synthesis.

Parameters:
p pointer to the AMRContext
lpc pointer to the LPC coefficients
fixed_gain fixed codebook gain for synthesis
fixed_vector algebraic codebook vector
samples pointer to the output speech samples
overflow 16-bit overflow flag

Definition at line 769 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), and amrwb_decode_frame().

static float tilt_factor ( float *  lpc_n,
float *  lpc_d 
) [static]

Get the tilt factor of a formant filter from its transfer function.

Parameters:
lpc_n LP_FILTER_ORDER coefficients of the numerator
lpc_d LP_FILTER_ORDER coefficients of the denominator

Definition at line 851 of file amrnbdec.c.

Referenced by calc_input_response(), postfilter(), and wiener_denoise().

static enum Mode unpack_bitstream ( AMRContext p,
const uint8_t buf,
int  buf_size 
) [static]

Unpack an RFC4867 speech frame into the AMR frame mode and parameters.

The order of speech bits is specified by 3GPP TS 26.101.

Parameters:
p the context
buf pointer to the input buffer
buf_size size of the input buffer
Returns:
the frame mode

Definition at line 188 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), and g723_1_decode_frame().

static void update_state ( AMRContext p  )  [static]

Update buffers and history at the end of decoding a subframe.

Parameters:
p pointer to the AMRContext

Definition at line 825 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

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 
) [static]

Double version of ff_weighted_vector_sumf().

Definition at line 141 of file amrnbdec.c.

Referenced by lsf2lsp_5().


Variable Documentation

Initial value:

 {
    .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 },
}

Definition at line 1058 of file amrnbdec.c.


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