libavcodec/amrwbdec.c File Reference

AMR wideband decoder. More...

#include "libavutil/lfg.h"
#include "avcodec.h"
#include "get_bits.h"
#include "lsp.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "amr.h"
#include "amrwbdata.h"

Go to the source code of this file.

Data Structures

struct  AMRWBContext

Defines

#define AMR_USE_16BIT_TABLES
#define BIT_STR(x, lsb, len)   (((x) >> (lsb)) & ((1 << (len)) - 1))
 Get x bits in the index interval [lsb,lsb+len-1] inclusive.
#define BIT_POS(x, p)   (((x) >> (p)) & 1)
 Get the bit at specified position.

Functions

static av_cold int amrwb_decode_init (AVCodecContext *avctx)
static int decode_mime_header (AMRWBContext *ctx, const uint8_t *buf)
 Decode the frame header in the "MIME/storage" format.
static void decode_isf_indices_36b (uint16_t *ind, float *isf_q)
 Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
static void decode_isf_indices_46b (uint16_t *ind, float *isf_q)
 Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
static void isf_add_mean_and_past (float *isf_q, float *isf_past)
 Apply mean and past ISF values using the prediction factor.
static void interpolate_isp (double isp_q[4][LP_ORDER], const double *isp4_past)
 Interpolate the fourth ISP vector from current and past frames to obtain an ISP vector for each subframe.
static void decode_pitch_lag_high (int *lag_int, int *lag_frac, int pitch_index, uint8_t *base_lag_int, int subframe)
 Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
static void decode_pitch_lag_low (int *lag_int, int *lag_frac, int pitch_index, uint8_t *base_lag_int, int subframe, enum Mode mode)
 Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
static void decode_pitch_vector (AMRWBContext *ctx, const AMRWBSubFrame *amr_subframe, const int subframe)
 Find the pitch vector by interpolating the past excitation at the pitch delay, which is obtained in this function.
static void decode_1p_track (int *out, int code, int m, int off)
 The next six functions decode_[i]p_track decode exactly i pulses positions and amplitudes (-1 or 1) in a subframe track using an encoded pulse indexing (TS 26.190 section 5.8.2).
static void decode_2p_track (int *out, int code, int m, int off)
 code: 2m+1 bits
static void decode_3p_track (int *out, int code, int m, int off)
 code: 3m+1 bits
static void decode_4p_track (int *out, int code, int m, int off)
 code: 4m bits
static void decode_5p_track (int *out, int code, int m, int off)
 code: 5m bits
static void decode_6p_track (int *out, int code, int m, int off)
 code: 6m-2 bits
static void decode_fixed_vector (float *fixed_vector, const uint16_t *pulse_hi, const uint16_t *pulse_lo, const enum Mode mode)
 Decode the algebraic codebook index to pulse positions and signs, then construct the algebraic codebook vector.
static void decode_gains (const uint8_t vq_gain, const enum Mode mode, float *fixed_gain_factor, float *pitch_gain)
 Decode pitch gain and fixed gain correction factor.
static void pitch_sharpening (AMRWBContext *ctx, float *fixed_vector)
 Apply pitch sharpening filters to the fixed codebook vector.
static float voice_factor (float *p_vector, float p_gain, float *f_vector, float f_gain)
 Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
static float * anti_sparseness (AMRWBContext *ctx, float *fixed_vector, float *buf)
 Reduce fixed vector sparseness by smoothing with one of three IR filters, also known as "adaptive phase dispersion".
static float stability_factor (const float *isf, const float *isf_past)
 Calculate a stability factor {teta} based on distance between current and past isf.
static float noise_enhancer (float fixed_gain, float *prev_tr_gain, float voice_fac, float stab_fac)
 Apply a non-linear fixed gain smoothing in order to reduce fluctuation in the energy of excitation.
static void pitch_enhancer (float *fixed_vector, float voice_fac)
 Filter the fixed_vector to emphasize the higher frequencies.
static void synthesis (AMRWBContext *ctx, float *lpc, float *excitation, float fixed_gain, const float *fixed_vector, float *samples)
 Conduct 16th order linear predictive coding synthesis from excitation.
static void de_emphasis (float *out, float *in, float m, float mem[1])
 Apply to synthesis a de-emphasis filter of the form: H(z) = 1 / (1 - m * z^-1).
static void upsample_5_4 (float *out, const float *in, int o_size)
 Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using a FIR interpolation filter.
static float find_hb_gain (AMRWBContext *ctx, const float *synth, uint16_t hb_idx, uint8_t vad)
 Calculate the high-band gain based on encoded index (23k85 mode) or on the low-band speech signal and the Voice Activity Detection flag.
static void scaled_hb_excitation (AMRWBContext *ctx, float *hb_exc, const float *synth_exc, float hb_gain)
 Generate the high-band excitation with the same energy from the lower one and scaled by the given gain.
static float auto_correlation (float *diff_isf, float mean, int lag)
 Calculate the auto-correlation for the ISF difference vector.
static void extrapolate_isf (float isf[LP_ORDER_16k])
 Extrapolate a ISF vector to the 16kHz range (20th order LP) used at mode 6k60 LP filter for the high frequency band.
static void lpc_weighting (float *out, const float *lpc, float gamma, int size)
 Spectral expand the LP coefficients using the equation: y[i] = x[i] * (gamma ** i).
static void hb_synthesis (AMRWBContext *ctx, int subframe, float *samples, const float *exc, const float *isf, const float *isf_past)
 Conduct 20th order linear predictive coding synthesis for the high frequency band excitation at 16kHz.
static void hb_fir_filter (float *out, const float fir_coef[HB_FIR_SIZE+1], float mem[HB_FIR_SIZE], const float *in)
 Apply a 15th order filter to high-band samples.
static void update_sub_state (AMRWBContext *ctx)
 Update context state before the next subframe.
static int amrwb_decode_frame (AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)

Variables

AVCodec ff_amrwb_decoder


Detailed Description

AMR wideband decoder.

Definition in file amrwbdec.c.


Define Documentation

#define AMR_USE_16BIT_TABLES

Definition at line 38 of file amrwbdec.c.

#define BIT_POS ( x,
 )     (((x) >> (p)) & 1)

Get the bit at specified position.

Definition at line 349 of file amrwbdec.c.

Referenced by decode_1p_track(), decode_2p_track(), decode_3p_track(), decode_4p_track(), decode_5p_track(), and decode_6p_track().

#define BIT_STR ( x,
lsb,
len   )     (((x) >> (lsb)) & ((1 << (len)) - 1))

Get x bits in the index interval [lsb,lsb+len-1] inclusive.

Definition at line 346 of file amrwbdec.c.

Referenced by decode_1p_track(), decode_2p_track(), decode_3p_track(), decode_4p_track(), decode_5p_track(), and decode_6p_track().


Function Documentation

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

Definition at line 1068 of file amrwbdec.c.

static av_cold int amrwb_decode_init ( AVCodecContext avctx  )  [static]

Definition at line 88 of file amrwbdec.c.

static float* anti_sparseness ( AMRWBContext ctx,
float *  fixed_vector,
float *  buf 
) [static]

Reduce fixed vector sparseness by smoothing with one of three IR filters, also known as "adaptive phase dispersion".

Parameters:
[in] ctx The context
[in,out] fixed_vector Unfiltered fixed vector
[out] buf Space for modified vector if necessary
Returns:
The potentially overwritten filtered fixed vector address

Definition at line 611 of file amrwbdec.c.

static float auto_correlation ( float *  diff_isf,
float  mean,
int  lag 
) [static]

Calculate the auto-correlation for the ISF difference vector.

Definition at line 885 of file amrwbdec.c.

Referenced by extrapolate_isf().

static void de_emphasis ( float *  out,
float *  in,
float  m,
float  mem[1] 
) [static]

Apply to synthesis a de-emphasis filter of the form: H(z) = 1 / (1 - m * z^-1).

Parameters:
[out] out Output buffer
[in] in Input samples array with in[-1]
[in] m Filter coefficient
[in,out] mem State from last filtering

Definition at line 791 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void decode_1p_track ( int *  out,
int  code,
int  m,
int  off 
) [inline, static]

The next six functions decode_[i]p_track decode exactly i pulses positions and amplitudes (-1 or 1) in a subframe track using an encoded pulse indexing (TS 26.190 section 5.8.2).

The results are given in out[], in which a negative number means amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).

Parameters:
[out] out Output buffer (writes i elements)
[in] code Pulse index (no. of bits varies, see below)
[in] m (log2) Number of potential positions
[in] off Offset for decoded positions

code: m+1 bits

Definition at line 364 of file amrwbdec.c.

Referenced by decode_3p_track(), decode_4p_track(), decode_6p_track(), and decode_fixed_vector().

static void decode_2p_track ( int *  out,
int  code,
int  m,
int  off 
) [inline, static]

code: 2m+1 bits

Definition at line 371 of file amrwbdec.c.

Referenced by decode_3p_track(), decode_4p_track(), decode_5p_track(), decode_6p_track(), and decode_fixed_vector().

static void decode_3p_track ( int *  out,
int  code,
int  m,
int  off 
) [static]

code: 3m+1 bits

Definition at line 381 of file amrwbdec.c.

Referenced by decode_4p_track(), decode_5p_track(), decode_6p_track(), and decode_fixed_vector().

static void decode_4p_track ( int *  out,
int  code,
int  m,
int  off 
) [static]

code: 4m bits

Definition at line 390 of file amrwbdec.c.

Referenced by decode_6p_track(), and decode_fixed_vector().

static void decode_5p_track ( int *  out,
int  code,
int  m,
int  off 
) [static]

code: 5m bits

Definition at line 426 of file amrwbdec.c.

Referenced by decode_6p_track(), and decode_fixed_vector().

static void decode_6p_track ( int *  out,
int  code,
int  m,
int  off 
) [static]

code: 6m-2 bits

Definition at line 436 of file amrwbdec.c.

Referenced by decode_fixed_vector().

static void decode_fixed_vector ( float *  fixed_vector,
const uint16_t *  pulse_hi,
const uint16_t *  pulse_lo,
const enum Mode  mode 
) [static]

Decode the algebraic codebook index to pulse positions and signs, then construct the algebraic codebook vector.

Parameters:
[out] fixed_vector Buffer for the fixed codebook excitation
[in] pulse_hi MSBs part of the pulse index array (higher modes only)
[in] pulse_lo LSBs part of the pulse index array
[in] mode Mode of the current frame

Definition at line 480 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void decode_gains ( const uint8_t  vq_gain,
const enum Mode  mode,
float *  fixed_gain_factor,
float *  pitch_gain 
) [static]

Decode pitch gain and fixed gain correction factor.

Parameters:
[in] vq_gain Vector-quantized index for gains
[in] mode Mode of the current frame
[out] fixed_gain_factor Decoded fixed gain correction factor
[out] pitch_gain Decoded pitch gain

Definition at line 551 of file amrwbdec.c.

static void decode_isf_indices_36b ( uint16_t *  ind,
float *  isf_q 
) [static]

Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).

Parameters:
[in] ind Array of 5 indexes
[out] isf_q Buffer for isf_q[LP_ORDER]

Definition at line 142 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void decode_isf_indices_46b ( uint16_t *  ind,
float *  isf_q 
) [static]

Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).

Parameters:
[in] ind Array of 7 indexes
[out] isf_q Buffer for isf_q[LP_ORDER]

Definition at line 169 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static int decode_mime_header ( AMRWBContext ctx,
const uint8_t *  buf 
) [static]

Decode the frame header in the "MIME/storage" format.

This format is simpler and does not carry the auxiliary frame information.

Parameters:
[in] ctx The Context
[in] buf Pointer to the input buffer
Returns:
The decoded header length in bytes

Definition at line 121 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void decode_pitch_lag_high ( int *  lag_int,
int *  lag_frac,
int  pitch_index,
uint8_t *  base_lag_int,
int  subframe 
) [static]

Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).

Calculate integer lag and fractional lag always using 1/4 resolution. In 1st and 3rd subframes the index is relative to last subframe integer lag.

Parameters:
[out] lag_int Decoded integer pitch lag
[out] lag_frac Decoded fractional pitch lag
[in] pitch_index Adaptive codebook pitch index
[in,out] base_lag_int Base integer lag used in relative subframes
[in] subframe Current subframe index (0 to 3)

Definition at line 245 of file amrwbdec.c.

Referenced by decode_pitch_vector().

static void decode_pitch_lag_low ( int *  lag_int,
int *  lag_frac,
int  pitch_index,
uint8_t *  base_lag_int,
int  subframe,
enum Mode  mode 
) [static]

Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.

The description is analogous to decode_pitch_lag_high, but in 6k60 the relative index is used for all subframes except the first.

Definition at line 278 of file amrwbdec.c.

Referenced by decode_pitch_vector().

static void decode_pitch_vector ( AMRWBContext ctx,
const AMRWBSubFrame amr_subframe,
const int  subframe 
) [static]

Find the pitch vector by interpolating the past excitation at the pitch delay, which is obtained in this function.

Parameters:
[in,out] ctx The context
[in] amr_subframe Current subframe data
[in] subframe Current subframe index (0 to 3)

Definition at line 307 of file amrwbdec.c.

static void extrapolate_isf ( float  isf[LP_ORDER_16k]  )  [static]

Extrapolate a ISF vector to the 16kHz range (20th order LP) used at mode 6k60 LP filter for the high frequency band.

Parameters:
[out] isf Buffer for extrapolated isf; contains LP_ORDER values on input

Definition at line 904 of file amrwbdec.c.

Referenced by hb_synthesis().

static float find_hb_gain ( AMRWBContext ctx,
const float *  synth,
uint16_t  hb_idx,
uint8_t  vad 
) [static]

Calculate the high-band gain based on encoded index (23k85 mode) or on the low-band speech signal and the Voice Activity Detection flag.

Parameters:
[in] ctx The context
[in] synth LB speech synthesis at 12.8k
[in] hb_idx Gain index for mode 23k85 only
[in] vad VAD flag for the frame

Definition at line 842 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void hb_fir_filter ( float *  out,
const float  fir_coef[HB_FIR_SIZE+1],
float  mem[HB_FIR_SIZE],
const float *  in 
) [static]

Apply a 15th order filter to high-band samples.

The filter characteristic depends on the given coefficients.

Parameters:
[out] out Buffer for filtered output
[in] fir_coef Filter coefficients
[in,out] mem State from last filtering (updated)
[in] in Input speech data (high-band)
Remarks:
It is safe to pass the same array in in and out parameters

Definition at line 1031 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void hb_synthesis ( AMRWBContext ctx,
int  subframe,
float *  samples,
const float *  exc,
const float *  isf,
const float *  isf_past 
) [static]

Conduct 20th order linear predictive coding synthesis for the high frequency band excitation at 16kHz.

Parameters:
[in] ctx The context
[in] subframe Current subframe index (0 to 3)
[in,out] samples Pointer to the output speech samples
[in] exc Generated white-noise scaled excitation
[in] isf Current frame isf vector
[in] isf_past Past frame final isf vector

Definition at line 992 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void interpolate_isp ( double  isp_q[4][LP_ORDER],
const double *  isp4_past 
) [static]

Interpolate the fourth ISP vector from current and past frames to obtain an ISP vector for each subframe.

Parameters:
[in,out] isp_q ISPs for each subframe
[in] isp4_past Past ISP for subframe 4

Definition at line 223 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void isf_add_mean_and_past ( float *  isf_q,
float *  isf_past 
) [static]

Apply mean and past ISF values using the prediction factor.

Updates past ISF vector.

Parameters:
[in,out] isf_q Current quantized ISF
[in,out] isf_past Past quantized ISF

Definition at line 203 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void lpc_weighting ( float *  out,
const float *  lpc,
float  gamma,
int  size 
) [static]

Spectral expand the LP coefficients using the equation: y[i] = x[i] * (gamma ** i).

Parameters:
[out] out Output buffer (may use input array)
[in] lpc LP coefficients array
[in] gamma Weighting factor
[in] size LP array size

Definition at line 970 of file amrwbdec.c.

Referenced by hb_synthesis().

static float noise_enhancer ( float  fixed_gain,
float *  prev_tr_gain,
float  voice_fac,
float  stab_fac 
) [static]

Apply a non-linear fixed gain smoothing in order to reduce fluctuation in the energy of excitation.

Parameters:
[in] fixed_gain Unsmoothed fixed gain
[in,out] prev_tr_gain Previous threshold gain (updated)
[in] voice_fac Frame voicing factor
[in] stab_fac Frame stability factor
Returns:
The smoothed gain

Definition at line 699 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void pitch_enhancer ( float *  fixed_vector,
float  voice_fac 
) [static]

Filter the fixed_vector to emphasize the higher frequencies.

Parameters:
[in,out] fixed_vector Fixed codebook vector
[in] voice_fac Frame voicing factor

Definition at line 726 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void pitch_sharpening ( AMRWBContext ctx,
float *  fixed_vector 
) [static]

Apply pitch sharpening filters to the fixed codebook vector.

Parameters:
[in] ctx The context
[in,out] fixed_vector Fixed codebook excitation

Definition at line 569 of file amrwbdec.c.

static void scaled_hb_excitation ( AMRWBContext ctx,
float *  hb_exc,
const float *  synth_exc,
float  hb_gain 
) [static]

Generate the high-band excitation with the same energy from the lower one and scaled by the given gain.

Parameters:
[in] ctx The context
[out] hb_exc Buffer for the excitation
[in] synth_exc Low-band excitation used for synthesis
[in] hb_gain Wanted excitation gain

Definition at line 867 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static float stability_factor ( const float *  isf,
const float *  isf_past 
) [static]

Calculate a stability factor {teta} based on distance between current and past isf.

A value of 1 shows maximum signal stability.

Definition at line 675 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void synthesis ( AMRWBContext ctx,
float *  lpc,
float *  excitation,
float  fixed_gain,
const float *  fixed_vector,
float *  samples 
) [static]

Conduct 16th order linear predictive coding synthesis from excitation.

Parameters:
[in] ctx Pointer to the AMRWBContext
[in] lpc Pointer to the LPC coefficients
[out] excitation Buffer for synthesis final excitation
[in] fixed_gain Fixed codebook gain for synthesis
[in] fixed_vector Algebraic codebook vector
[in,out] samples Pointer to the output samples and memory

Definition at line 754 of file amrwbdec.c.

static void update_sub_state ( AMRWBContext ctx  )  [static]

Update context state before the next subframe.

Definition at line 1052 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static void upsample_5_4 ( float *  out,
const float *  in,
int  o_size 
) [static]

Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using a FIR interpolation filter.

Uses past data from before *in address.

Parameters:
[out] out Buffer for interpolated signal
[in] in Current signal data (length 0.8*o_size)
[in] o_size Output signal length

Definition at line 811 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

static float voice_factor ( float *  p_vector,
float  p_gain,
float *  f_vector,
float  f_gain 
) [static]

Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).

Parameters:
[in] p_vector,f_vector Pitch and fixed excitation vectors
[in] p_gain,f_gain Pitch and fixed gains

Definition at line 590 of file amrwbdec.c.

Referenced by amrwb_decode_frame().


Variable Documentation

Initial value:

 {
    .name           = "amrwb",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = CODEC_ID_AMR_WB,
    .priv_data_size = sizeof(AMRWBContext),
    .init           = amrwb_decode_init,
    .decode         = amrwb_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
    .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"),
    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
}

Definition at line 1243 of file amrwbdec.c.


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