Libav 0.7.1
Data Structures | Defines | Typedefs | Functions | Variables
libavcodec/amrnbdec.c File Reference

AMR narrowband decoder. More...

#include <string.h>
#include <math.h>
#include "avcodec.h"
#include "get_bits.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.

Typedefs

typedef struct AMRContext AMRContext

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 *data_size, 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 95 of file amrnbdec.c.

Referenced by postfilter().

#define AMR_BLOCK_SIZE   160

samples per frame

Definition at line 59 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

#define AMR_SAMPLE_BOUND   32768.0

threshold for synthesis overflow

Definition at line 60 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 71 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 93 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 91 of file amrnbdec.c.

Referenced by tilt_factor().

#define LSF_R_FAC   (8000.0 / 32768.0)

LSF residual tables to Hertz.

Definition at line 76 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 81 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 77 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 78 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 74 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 88 of file amrnbdec.c.

Referenced by pitch_sharpening(), and synthesis().


Typedef Documentation

typedef struct AMRContext AMRContext

Function Documentation

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

Definition at line 922 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:
pthe context
fixed_sparsealgebraic codebook vector
fixed_vectorunfiltered fixed vector
fixed_gainsmoothed gain
outspace for modified vector if necessary

Definition at line 696 of file amrnbdec.c.

Referenced by amrnb_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:
outvector with filter applied
insource vector
filterphase 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 649 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 412 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_indexpositions of the eight pulses
fixed_sparsepointer to the algebraic codebook vector

Definition at line 430 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_sparsepointer to the algebraic codebook vector
pulsesalgebraic codebook indexes
modemode of the current frame
subframecurrent subframe number

Definition at line 476 of file amrnbdec.c.

Referenced by amrnb_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:
pthe context
amr_subframeunpacked amr subframe
modemode of the current frame
subframecurrent subframe number
fixed_gain_factordecoded gain correction factor

Definition at line 607 of file amrnbdec.c.

Referenced by amrnb_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 350 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 369 of file amrnbdec.c.

Referenced by amrnb_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:
pthe context
lsfLSFs for the current subframe, in the range [0,1]
lsf_avgaveraged LSFs
modemode of the current frame
Returns:
fixed gain smoothed

Definition at line 565 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_qLSFs in [0,1] for each subframe
[in]lsf_newNew LSFs in [0,1] for subframe 4

Definition at line 217 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:
ppointer to the AMRContext

Definition at line 305 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:
ppointer to the AMRContext

Definition at line 276 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:
pthe context
lspoutput LSP vector
lsf_no_rLSF vector without the residual vector added
lsf_quantizerpointers to LSF dictionary tables
quantizer_offsetoffset in tables
signfor the 3 dictionary table
updatestore data for computing the next frame's LSFs

Definition at line 238 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:
pthe context
subframeunpacked amr subframe
modemode of the current frame
fixed_sparsesparse respresentation of the fixed vector

Definition at line 529 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

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:
ppointer to the AMRContext
lpcinterpolated LP coefficients for this subframe
buf_outoutput of the filter

Definition at line 878 of file amrnbdec.c.

Referenced by amrnb_decode_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:
ppointer to the AMRContext
lpcpointer to the LPC coefficients
fixed_gainfixed codebook gain for synthesis
fixed_vectoralgebraic codebook vector
samplespointer to the output speech samples
overflow16-bit overflow flag

Definition at line 767 of file amrnbdec.c.

Referenced by amrnb_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_nLP_FILTER_ORDER coefficients of the numerator
lpc_dLP_FILTER_ORDER coefficients of the denominator

Definition at line 849 of file amrnbdec.c.

Referenced by postfilter().

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:
pthe context
bufpointer to the input buffer
buf_sizesize of the input buffer
Returns:
the frame mode

Definition at line 185 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

static void update_state ( AMRContext p) [static]

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

Parameters:
ppointer to the AMRContext

Definition at line 823 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,
    .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 1039 of file amrnbdec.c.