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)
	Decodes quantized ISF vectors using 36-bit indexes (6K60 mode only).
static void	decode_isf_indices_46b (uint16_t *ind, float *isf_q)
	Decodes 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 Updates past ISF vector.
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 a 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) 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.
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 a adaptive codebook index into pitch lag for 8k85 and 6k60 modes Description is analogous to decode_pitch_lag_high, but in 6k60 relative index is used for all subframes except the first.
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 out[LP_ORDER_16k], float isf[LP_ORDER])
	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 to high-band samples a 15th order filter The filter characteristic depends on the given coefficients.
static void	update_sub_state (AMRWBContext *ctx)
	Update context state before the next subframe.
static int	amrwb_decode_frame (AVCodecContext *avctx, void *data, int *data_size, 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,
		p		)	(((x) >> (p)) & 1)

Get the bit at specified position.

Definition at line 345 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 342 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 *	data_size,
		AVPacket *	avpkt
	)			[static]

Definition at line 1065 of file amrwbdec.c.

static av_cold int amrwb_decode_init

(

AVCodecContext *

avctx

)

[static]

Definition at line 87 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 607 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 881 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 787 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 360 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 367 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 377 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 386 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 422 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 432 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 476 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 547 of file amrwbdec.c.

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

Decodes 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 138 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

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

Decodes 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 165 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 information of the frame

Parameters:

[in]	ctx	The Context
[in]	buf	Pointer to the input buffer

Returns:

The decoded header length in bytes

Definition at line 117 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 241 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 a adaptive codebook index into pitch lag for 8k85 and 6k60 modes Description is analogous to decode_pitch_lag_high, but in 6k60 relative index is used for all subframes except the first.

Definition at line 274 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 303 of file amrwbdec.c.

static void extrapolate_isf	(	float	out[LP_ORDER_16k],
		float	isf[LP_ORDER]
	)			[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]	out	Buffer for extrapolated isf
[in]	isf	Input isf vector

Definition at line 900 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 838 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 to high-band samples a 15th order filter 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 1028 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 989 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 a 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 219 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 199 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 967 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 695 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 722 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 565 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 863 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 671 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 750 of file amrwbdec.c.

static void update_sub_state

(

AMRWBContext *

ctx

)

[static]

Update context state before the next subframe.

Definition at line 1049 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 807 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 586 of file amrwbdec.c.

Referenced by amrwb_decode_frame().

---

Variable Documentation

AVCodec ff_amrwb_decoder

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,
    .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 1228 of file amrwbdec.c.