libavcodec/amrnbdec.c File Reference

AMR narrowband decoder. More...

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

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 (ACELPVContext *ctx, 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 (AMRContext *p, 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.

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.

Definition in file amrnbdec.c.

Define Documentation

#define AMR_AGC_ALPHA 0.9

Adaptive gain control factor used in post-filter.

Definition at line 96 of file amrnbdec.c.

Referenced by postfilter().

#define AMR_BLOCK_SIZE 160

samples per frame

Definition at line 60 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

#define AMR_SAMPLE_BOUND 32768.0

threshold for synthesis overflow

Definition at line 61 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 72 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 94 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 92 of file amrnbdec.c.

Referenced by tilt_factor().

#define LSF_R_FAC (8000.0 / 32768.0)

LSF residual tables to Hertz.

Definition at line 77 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 82 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 78 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 79 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 75 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 89 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 941 of file amrnbdec.c.

static av_cold int amrnb_decode_init ( AVCodecContext * avctx ) [static]

Definition at line 159 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 712 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 665 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 428 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 446 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.

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 492 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 623 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 365 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 384 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 581 of file amrnbdec.c.

Referenced by amrnb_decode_frame().

static void interpolate_lsf	(	ACELPVContext *	ctx,
		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]`	ctx	The Context
`[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 232 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 320 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 291 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 253 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 545 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 897 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 783 of file amrnbdec.c.

Referenced by amrnb_decode_frame(), and amrwb_decode_frame().

static float tilt_factor	(	AMRContext *	p,
		float *	lpc_n,
		float *	lpc_d
	)			`[static]`

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

Parameters:

	p	The Context
	lpc_n	LP_FILTER_ORDER coefficients of the numerator
	lpc_d	LP_FILTER_ORDER coefficients of the denominator

Definition at line 867 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 200 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 840 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 148 of file amrnbdec.c.

Referenced by lsf2lsp_5().

Variable Documentation

AVCodec ff_amrnb_decoder

Initial value:

 {
    .name           = "amrnb",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_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("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
                                                     AV_SAMPLE_FMT_NONE },
}

Definition at line 1078 of file amrnbdec.c.


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 (ACELPVContext ctx, 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 (AMRContext p, 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