60 #define AMR_BLOCK_SIZE 160
61 #define AMR_SAMPLE_BOUND 32768.0
72 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)
75 #define PRED_FAC_MODE_12k2 0.65
77 #define LSF_R_FAC (8000.0 / 32768.0)
78 #define MIN_LSF_SPACING (50.0488 / 8000.0)
79 #define PITCH_LAG_MIN_MODE_12k2 18
82 #define MIN_ENERGY -14.0
89 #define SHARP_MAX 0.79449462890625
92 #define AMR_TILT_RESPONSE 22
94 #define AMR_TILT_GAMMA_T 0.8
96 #define AMR_AGC_ALPHA 0.9
144 const double *in_b,
double weight_coeff_a,
145 double weight_coeff_b,
int length)
149 for (i = 0; i < length; i++)
150 out[i] = weight_coeff_a * in_a[i]
151 + weight_coeff_b * in_b[i];
177 for (i = 0; i < 4; i++)
204 mode = buf[0] >> 3 & 0x0F;
233 for (i = 0; i < 4; i++)
235 0.25 * (3 - i), 0.25 * (i + 1),
251 const float lsf_no_r[LP_FILTER_ORDER],
252 const int16_t *lsf_quantizer[5],
253 const int quantizer_offset,
254 const int sign,
const int update)
260 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
261 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
270 memcpy(p->
prev_lsf_r, lsf_r, LP_FILTER_ORDER *
sizeof(*lsf_r));
273 lsf_q[i] = lsf_r[i] * (
LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
290 const uint16_t *lsf_param = p->
frame.
lsf;
292 const int16_t *lsf_quantizer[5];
295 lsf_quantizer[0] =
lsf_5_1[lsf_param[0]];
296 lsf_quantizer[1] =
lsf_5_2[lsf_param[1]];
297 lsf_quantizer[2] =
lsf_5_3[lsf_param[2] >> 1];
298 lsf_quantizer[3] =
lsf_5_4[lsf_param[3]];
299 lsf_quantizer[4] =
lsf_5_5[lsf_param[4]];
319 const uint16_t *lsf_param = p->
frame.
lsf;
322 const int16_t *lsf_quantizer;
326 memcpy(lsf_r, lsf_quantizer, 3 *
sizeof(*lsf_r));
329 memcpy(lsf_r + 3, lsf_quantizer, 3 *
sizeof(*lsf_r));
332 memcpy(lsf_r + 6, lsf_quantizer, 4 *
sizeof(*lsf_r));
342 memcpy(p->
prev_lsf_r, lsf_r, LP_FILTER_ORDER *
sizeof(*lsf_r));
347 for (i = 1; i <= 3; i++)
363 const int prev_lag_int,
const int subframe)
365 if (subframe == 0 || subframe == 2) {
366 if (pitch_index < 463) {
367 *lag_int = (pitch_index + 107) * 10923 >> 16;
368 *lag_frac = pitch_index - *lag_int * 6 + 105;
370 *lag_int = pitch_index - 368;
374 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
375 *lag_frac = pitch_index - *lag_int * 6 - 3;
385 int pitch_lag_int, pitch_lag_frac;
403 pitch_lag_int += pitch_lag_frac > 0;
409 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
425 int i1,
int i2,
int i3)
430 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
431 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
432 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
445 int pulse_position[8];
453 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
454 pulse_position[3] = temp % 5;
455 pulse_position[7] = temp / 5;
456 if (pulse_position[7] & 1)
457 pulse_position[3] = 4 - pulse_position[3];
458 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
459 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
462 for (i = 0; i < 4; i++) {
463 const int pos1 = (pulse_position[i] << 2) + i;
464 const int pos2 = (pulse_position[i + 4] << 2) + i;
465 const float sign = fixed_index[i] ? -1.0 : 1.0;
466 fixed_sparse->
x[i ] = pos1;
467 fixed_sparse->
x[i + 4] = pos2;
468 fixed_sparse->
y[i ] = sign;
469 fixed_sparse->
y[i + 4] = pos2 < pos1 ? -sign : sign;
489 const enum Mode mode,
const int subframe)
498 int *pulse_position = fixed_sparse->
x;
500 const int fixed_index = pulses[0];
503 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
504 pulse_position[0] = ( fixed_index & 7) * 5 +
track_position[pulse_subset];
505 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 +
track_position[pulse_subset + 1];
508 pulse_subset = ((fixed_index & 1) << 1) + 1;
509 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
510 pulse_subset = (fixed_index >> 4) & 3;
511 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
512 fixed_sparse->
n = pulse_position[0] == pulse_position[1] ? 1 : 2;
514 pulse_position[0] = (fixed_index & 7) * 5;
515 pulse_subset = (fixed_index >> 2) & 2;
516 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
517 pulse_subset = (fixed_index >> 6) & 2;
518 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
522 pulse_position[1] =
gray_decode[(fixed_index >> 3) & 7] + 1;
523 pulse_position[2] =
gray_decode[(fixed_index >> 6) & 7] + 2;
524 pulse_subset = (fixed_index >> 9) & 1;
525 pulse_position[3] =
gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
528 for (i = 0; i < fixed_sparse->
n; i++)
529 fixed_sparse->
y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
578 const float *lsf_avg,
const enum Mode mode)
584 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
600 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
605 (1.0 - smoothing_factor) * fixed_gain_mean;
620 const enum Mode mode,
const int subframe,
621 float *fixed_gain_factor)
629 const uint16_t *gains;
640 p->
pitch_gain[4] = gains[0] * (1.0 / 16384.0);
641 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
674 if (lag < AMR_SUBFRAME_SIZE >> 1)
680 for (i = 0; i < in->
n; i++) {
683 const float *filterp;
685 if (x >= AMR_SUBFRAME_SIZE - lag) {
687 }
else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
709 const float *fixed_vector,
710 float fixed_gain,
float *out)
730 for (i = 0; i < 5; i++)
738 }
else if (ir_filter_nr < 2)
744 if (fixed_gain < 5.0)
748 && ir_filter_nr < 2) {
780 float fixed_gain,
const float *fixed_vector,
793 p->
pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
870 memcpy(hf + 1, lpc_n,
sizeof(
float) * LP_FILTER_ORDER);
899 const float *gamma_n, *gamma_d;
911 lpc_n[i] = lpc[i] * gamma_n[i];
912 lpc_d[i] = lpc[i] * gamma_d[i];
915 memcpy(pole_out, p->
postfilter_mem,
sizeof(
float) * LP_FILTER_ORDER);
919 sizeof(
float) * LP_FILTER_ORDER);
922 pole_out + LP_FILTER_ORDER,
935 int *got_frame_ptr,
AVPacket *avpkt)
940 int buf_size = avpkt->
size;
942 int i, subframe, ret;
943 float fixed_gain_factor;
946 float synth_fixed_gain;
947 const float *synth_fixed_vector;
972 for (i = 0; i < 4; i++)
975 for (subframe = 0; subframe < 4; subframe++) {
1030 synth_fixed_gain, spare_vector);
1040 postfilter(p, p->
lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
#define AMR_SAMPLE_SCALE
Scale from constructed speech to [-1,1].
void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index, const int prev_lag_int, const int subframe, int third_as_first, int resolution)
Decode the adaptive codebook index to the integer and fractional parts of the pitch lag for one subfr...
#define AMR_BLOCK_SIZE
samples per frame
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
This structure describes decoded (raw) audio or video data.
float lsf_avg[LP_FILTER_ORDER]
vector of averaged lsf vector
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
void ff_decode_10_pulses_35bits(const int16_t *fixed_index, AMRFixed *fixed_sparse, const uint8_t *gray_decode, int half_pulse_count, int bits)
Decode the algebraic codebook index to pulse positions and signs and construct the algebraic codebook...
static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
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)
AVFrame avframe
AVFrame for decoded samples.
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length)
float implementation of weighted sum of two vectors.
AVFrame * coded_frame
the picture in the bitstream
AMRNB unpacked data frame.
void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
Clear array values set by set_fixed_vector.
static const uint8_t base_five_table[128][3]
Base-5 representation for values 0-124.
static const int16_t lsf_3_1[256][3]
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 codebo...
static float tilt_factor(float *lpc_n, float *lpc_d)
Get the tilt factor of a formant filter from its transfer function.
static const uint8_t track_position[16]
track start positions for algebraic code book routines
uint8_t bad_frame_indicator
bad frame ? 1 : 0
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
Add fixed vector to an array from a sparse representation.
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...
static const int16_t lsf_3_2[512][3]
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.
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 const float * ir_filters_lookup_MODE_7k95[2]
static av_cold int amrnb_decode_init(AVCodecContext *avctx)
double prev_lsp_sub4[LP_FILTER_ORDER]
lsp vector for the 4th subframe of the previous frame
static void postfilter(AMRContext *p, float *lpc, float *buf_out)
Perform adaptive post-filtering to enhance the quality of the speech.
static const int16_t lsf_5_1[128][4]
float postfilter_agc
previous factor used for adaptive gain control
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
enum AVSampleFormat sample_fmt
audio sample format
Sparse representation for the algebraic codebook (fixed) vector.
static const uint16_t qua_gain_code[32]
scalar quantized fixed gain table for 7.95 and 12.2 kbps modes
Mode
Frame type (Table 1a in 3GPP TS 26.101)
static const uint16_t qua_gain_pit[16]
scalar quantized pitch gain table for 7.95 and 12.2 kbps modes
static void lsf2lsp_3(AMRContext *p)
Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
static const float energy_pred_fac[4]
4-tap moving average prediction coefficients in reverse order
uint16_t fixed_gain
index to decode the fixed gain factor, for MODE_12k2 and MODE_7k95
static const int8_t lsp_sub4_init[LP_FILTER_ORDER]
Values for the lsp vector from the 4th subframe of the previous subframe values.
double lsp[4][LP_FILTER_ORDER]
lsp vectors from current frame
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...
AMRNBFrame frame
decoded AMR parameters (lsf coefficients, codebook indexes, etc)
static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
Interpolate the LSF vector (used for fixed gain smoothing).
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ, int size, float alpha, float *gain_mem)
Adaptive gain control (as used in AMR postfiltering)
static void ff_amr_bit_reorder(uint16_t *out, int size, const uint8_t *data, const R_TABLE_TYPE *ord_table)
Fill the frame structure variables from bitstream by parsing the given reordering table that uses the...
uint16_t lsf[5]
lsf parameters: 5 parameters for MODE_12k2, only 3 for other modes
static int init(AVCodecParserContext *s)
uint8_t prev_ir_filter_nr
previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
static void decode_pitch_vector(AMRContext *p, const AMRNBSubframe *amr_subframe, const int subframe)
static void update_state(AMRContext *p)
Update buffers and history at the end of decoding a subframe.
float fixed_vector[AMR_SUBFRAME_SIZE]
algebraic codebook (fixed) vector (must be kept zero between frames)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
AMRNBSubframe subframe[4]
unpacked data for each subframe
const float ff_pow_0_7[10]
Table of pow(0.7,n)
void av_log(void *avcl, int level, const char *fmt,...)
float ff_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
const char * name
Name of the codec implementation.
int16_t prev_lsf_r[LP_FILTER_ORDER]
residual LSF vector from previous subframe
static const uint8_t frame_sizes_nb[N_MODES]
number of bytes for each mode
void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in, float sum_of_squares, const int n)
Set the sum of squares of a signal by scaling.
const float ff_pow_0_75[10]
Table of pow(0.75,n)
float pitch_gain[5]
quantified pitch gains for the current and previous four subframes
#define LP_FILTER_ORDER
linear predictive coding filter order
static const int16_t lsf_3_3_MODE_5k15[128][4]
float * excitation
pointer to the current excitation vector in excitation_buf
uint64_t channel_layout
Audio channel layout.
#define AMR_SAMPLE_BOUND
threshold for synthesis overflow
uint8_t ir_filter_onset
flag for impulse response filter strength
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
#define AMR_SUBFRAME_SIZE
samples per subframe
AMRNB unpacked data subframe.
audio channel layout utility functions
#define MIN_ENERGY
Initial energy in dB.
static const float highpass_poles[2]
float samples_in[LP_FILTER_ORDER+AMR_SUBFRAME_SIZE]
floating point samples
static const int16_t lsf_3_1_MODE_7k95[512][3]
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame)
Get a buffer for a frame.
static const int16_t lsf_5_5[64][4]
static const float * ir_filters_lookup[2]
uint16_t p_lag
index to decode the pitch lag
static av_always_inline av_const float truncf(float x)
static const float highpass_zeros[2]
static const uint16_t gains_MODE_4k75[512][2]
gain table for 4.75 kbps mode
float pitch_vector[AMR_SUBFRAME_SIZE]
adaptive code book (pitch) vector
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
#define MIN_LSF_SPACING
Ensures stability of LPC filter.
static const float lsf_3_mean[LP_FILTER_ORDER]
void ff_acelp_lspd2lpc(const double *lsp, float *lpc, int lp_half_order)
Reconstruct LPC coefficients from the line spectral pair frequencies.
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.
uint8_t pitch_lag_int
integer part of pitch lag from current subframe
float tilt_mem
previous input to tilt compensation filter
float lsf_q[4][LP_FILTER_ORDER]
Interpolated LSF vector for fixed gain smoothing.
#define PRED_FAC_MODE_12k2
Prediction factor for 12.2kbit/s mode.
void ff_celp_circ_addf(float *out, const float *in, const float *lagged, int lag, float fac, int n)
Add an array to a rotated array.
int sample_rate
samples per second
float high_pass_mem[2]
previous intermediate values in the high-pass filter
main external API structure.
static const float lsf_5_mean[LP_FILTER_ORDER]
uint16_t p_gain
index to decode the pitch gain
uint8_t diff_count
the number of subframes for which diff has been above 0.65
static const uint8_t *const amr_unpacking_bitmaps_per_mode[N_MODES]
position of the bitmapping data for each packet type in the AMRNBFrame
void avcodec_get_frame_defaults(AVFrame *frame)
Set the fields of the given AVFrame to default values.
float prediction_error[4]
quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
void av_log_missing_feature(void *avc, const char *feature, int want_sample)
Log a generic warning message about a missing feature.
static const float highpass_gain
void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
Floating point version of ff_acelp_lsf2lsp()
float fixed_gain[5]
quantified fixed gains for the current and previous four subframes
void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP zero synthesis filter.
float lpc[4][LP_FILTER_ORDER]
lpc coefficient vectors for 4 subframes
float beta
previous pitch_gain, bounded by [0.0,SHARP_MAX]
#define SHARP_MAX
Maximum sharpening factor.
#define AMR_TILT_RESPONSE
Number of impulse response coefficients used for tilt factor.
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...
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 const int16_t lsf_5_4[256][4]
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 const int16_t lsf_3_3[512][4]
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static const int16_t lsf_5_2[256][4]
static const uint8_t gray_decode[8]
3-bit Gray code to binary lookup table
static const float pred_fac[LP_FILTER_ORDER]
Prediction factor table for modes other than 12.2kbit/s.
float prev_sparse_fixed_gain
previous fixed gain; used by anti-sparseness processing to determine "onset"
float postfilter_mem[10]
previous intermediate values in the formant filter
#define AMR_AGC_ALPHA
Adaptive gain control factor used in post-filter.
common internal api header.
common internal and external API header
static void lsf2lsp_5(AMRContext *p)
Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
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 const uint16_t gains_low[64][2]
gain table for 5.15 and 5.90 kbps modes
#define AVERROR_INVALIDDATA
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).
void ff_set_min_dist_lsf(float *lsf, double min_spacing, int size)
Adjust the quantized LSFs so they are increasing and not too close.
AVSampleFormat
Audio Sample Formats.
#define AVERROR_PATCHWELCOME
#define LSF_R_FAC
LSF residual tables to Hertz.
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
static const uint16_t gains_high[128][2]
gain table for 6.70, 7.40 and 10.2 kbps modes
uint8_t hang_count
the number of subframes since a hangover period started
int channels
number of audio channels
AMR narrowband data and definitions.
static const float energy_mean[8]
desired mean innovation energy, indexed by active mode
void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
#define PITCH_LAG_MIN_MODE_12k2
Lower bound on decoded lag search in 12.2kbit/s mode.
static const int8_t pulses[4]
uint16_t pulses[10]
pulses: 10 for MODE_12k2, 7 for MODE_10k2, and index and sign for others
struct AMRContext AMRContext
float excitation_buf[PITCH_DELAY_MAX+LP_FILTER_ORDER+1+AMR_SUBFRAME_SIZE]
current excitation and all necessary excitation history
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.
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
const float ff_pow_0_55[10]
Table of pow(0.55,n)
static const int16_t lsf_5_3[256][4]
#define AMR_TILT_GAMMA_T
Tilt factor = 1st reflection coefficient * gamma_t.
int nb_samples
number of audio samples (per channel) described by this frame
float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy, float *prediction_error, float energy_mean, const float *pred_table)
Calculate fixed gain (part of section 6.1.3 of AMR spec)
static const int16_t lsp_avg_init[LP_FILTER_ORDER]
Mean lsp values.