FFmpeg
4.0
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AMR narrowband decoder. More...
#include <string.h>
#include <math.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "avcodec.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 "internal.h"
#include "amrnbdata.h"
Go to the source code of this file.
Data Structures | |
struct | AMRContext |
Macros | |
#define | AMR_BLOCK_SIZE 160 |
samples per frame More... | |
#define | AMR_SAMPLE_BOUND 32768.0 |
threshold for synthesis overflow More... | |
#define | AMR_SAMPLE_SCALE (2.0 / 32768.0) |
Scale from constructed speech to [-1,1]. More... | |
#define | PRED_FAC_MODE_12k2 0.65 |
Prediction factor for 12.2kbit/s mode. More... | |
#define | LSF_R_FAC (8000.0 / 32768.0) |
LSF residual tables to Hertz. More... | |
#define | MIN_LSF_SPACING (50.0488 / 8000.0) |
Ensures stability of LPC filter. More... | |
#define | PITCH_LAG_MIN_MODE_12k2 18 |
Lower bound on decoded lag search in 12.2kbit/s mode. More... | |
#define | MIN_ENERGY -14.0 |
Initial energy in dB. More... | |
#define | SHARP_MAX 0.79449462890625 |
Maximum sharpening factor. More... | |
#define | AMR_TILT_RESPONSE 22 |
Number of impulse response coefficients used for tilt factor. More... | |
#define | AMR_TILT_GAMMA_T 0.8 |
Tilt factor = 1st reflection coefficient * gamma_t. More... | |
#define | AMR_AGC_ALPHA 0.9 |
Adaptive gain control factor used in post-filter. More... | |
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() More... | |
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. More... | |
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). More... | |
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. More... | |
static void | lsf2lsp_5 (AMRContext *p) |
Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors. More... | |
static void | lsf2lsp_3 (AMRContext *p) |
Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector. More... | |
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. More... | |
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. More... | |
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. More... | |
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. More... | |
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) More... | |
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. More... | |
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). More... | |
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). More... | |
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. More... | |
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. More... | |
AMR update functions | |
static void | update_state (AMRContext *p) |
Update buffers and history at the end of decoding a subframe. More... | |
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. More... | |
static void | postfilter (AMRContext *p, float *lpc, float *buf_out) |
Perform adaptive post-filtering to enhance the quality of the speech. More... | |
Variables | |
AVCodec | ff_amrnb_decoder |
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 AMR_BLOCK_SIZE 160 |
#define AMR_SAMPLE_BOUND 32768.0 |
threshold for synthesis overflow
Definition at line 63 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 74 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
#define PRED_FAC_MODE_12k2 0.65 |
Prediction factor for 12.2kbit/s mode.
Definition at line 77 of file amrnbdec.c.
Referenced by lsf2lsp_5().
#define LSF_R_FAC (8000.0 / 32768.0) |
LSF residual tables to Hertz.
Definition at line 79 of file amrnbdec.c.
Referenced by lsf2lsp_3(), lsf2lsp_5(), and lsf2lsp_for_mode12k2().
#define MIN_LSF_SPACING (50.0488 / 8000.0) |
Ensures stability of LPC filter.
Definition at line 80 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 81 of file amrnbdec.c.
Referenced by decode_pitch_lag_1_6().
#define MIN_ENERGY -14.0 |
Initial energy in dB.
Also used for bad frames (unimplemented).
Definition at line 84 of file amrnbdec.c.
Referenced by amrnb_decode_init(), and amrwb_decode_init().
#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 91 of file amrnbdec.c.
Referenced by pitch_sharpening(), and synthesis().
#define AMR_TILT_RESPONSE 22 |
Number of impulse response coefficients used for tilt factor.
Definition at line 94 of file amrnbdec.c.
Referenced by tilt_factor().
#define AMR_TILT_GAMMA_T 0.8 |
Tilt factor = 1st reflection coefficient * gamma_t.
Definition at line 96 of file amrnbdec.c.
Referenced by tilt_factor().
#define AMR_AGC_ALPHA 0.9 |
Adaptive gain control factor used in post-filter.
Definition at line 98 of file amrnbdec.c.
Referenced by postfilter().
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Double version of ff_weighted_vector_sumf()
Definition at line 149 of file amrnbdec.c.
Referenced by lsf2lsp_5().
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Definition at line 160 of file amrnbdec.c.
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Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
The order of speech bits is specified by 3GPP TS 26.101.
p | the context |
buf | pointer to the input buffer |
buf_size | size of the input buffer |
Definition at line 207 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Interpolate the LSF vector (used for fixed gain smoothing).
The interpolation is done over all four subframes even in MODE_12k2.
[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 239 of file amrnbdec.c.
Referenced by lsf2lsp_3(), and lsf2lsp_for_mode12k2().
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Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
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 260 of file amrnbdec.c.
Referenced by lsf2lsp_5().
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Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
p | pointer to the AMRContext |
Definition at line 298 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
p | pointer to the AMRContext |
Definition at line 327 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Like ff_decode_pitch_lag(), but with 1/6 resolution.
Definition at line 372 of file amrnbdec.c.
Referenced by decode_pitch_vector().
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Definition at line 391 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
Definition at line 435 of file amrnbdec.c.
Referenced by decode_8_pulses_31bits().
Decode the algebraic codebook index to pulse positions and signs and construct the algebraic codebook vector for MODE_10k2.
fixed_index | positions of the eight pulses |
fixed_sparse | pointer to the algebraic codebook vector |
Definition at line 453 of file amrnbdec.c.
Referenced by decode_fixed_sparse().
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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
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 499 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
p | the context |
subframe | unpacked amr subframe |
mode | mode of the current frame |
fixed_sparse | sparse representation of the fixed vector |
Definition at line 552 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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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.
p | the context |
lsf | LSFs for the current subframe, in the range [0,1] |
lsf_avg | averaged LSFs |
mode | mode of the current frame |
Definition at line 588 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Decode pitch gain and fixed gain factor (part of section 6.1.3).
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 630 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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).
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 672 of file amrnbdec.c.
Referenced by anti_sparseness().
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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).
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 719 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Conduct 10th order linear predictive coding synthesis.
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 790 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Update buffers and history at the end of decoding a subframe.
p | pointer to the AMRContext |
Definition at line 847 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Get the tilt factor of a formant filter from its transfer function.
p | The Context |
lpc_n | LP_FILTER_ORDER coefficients of the numerator |
lpc_d | LP_FILTER_ORDER coefficients of the denominator |
Definition at line 874 of file amrnbdec.c.
Referenced by postfilter().
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Perform adaptive post-filtering to enhance the quality of the speech.
See section 6.2.1.
p | pointer to the AMRContext |
lpc | interpolated LP coefficients for this subframe |
buf_out | output of the filter |
Definition at line 904 of file amrnbdec.c.
Referenced by amrnb_decode_frame().
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Definition at line 948 of file amrnbdec.c.
AVCodec ff_amrnb_decoder |
Definition at line 1083 of file amrnbdec.c.