AMR wideband decoder. More...

#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "lsp.h"
#include "celp_filters.h"
#include "celp_math.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "internal.h"
#include "amr.h"
#include "amrwbdata.h"
#include "mips/amrwbdec_mips.h"

Data Structures
struct	AMRWBContext

Macros
#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, CELPMContext *ctx)
	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, CELPMContext *ctx)
	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.

Macro Definition Documentation

#define AMR_USE_16BIT_TABLES

Definition at line 41 of file amrwbdec.c.

#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 361 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_POS	(	x,
		p
	)	(((x) >> (p)) & 1)

Get the bit at specified position.

Definition at line 364 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 av_cold int amrwb_decode_init ( AVCodecContext * avctx )

static

Definition at line 96 of file amrwbdec.c.

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 140 of file amrwbdec.c.

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

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

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

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

[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)

[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

[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

[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

[in]	p_vector,f_vector	Pitch and fixed excitation vectors
[in]	p_gain,f_gain	Pitch and fixed gains
[in]	ctx	The context

[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

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

[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

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

[out]	out	Buffer for interpolated signal
[in]	in	Current signal data (length 0.8*o_size)
[in]	o_size	Output signal length
[in]	ctx	The context

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

[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)

Data Structures

Macros

Functions

Variables

Detailed Description

Macro Definition Documentation

Function Documentation

Variable Documentation