76 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
100 0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
101 0.117649, 0.082354, 0.057648, 0.040354, 0.028248
105 0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
106 0.177979, 0.133484, 0.100113, 0.075085, 0.056314
110 0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
111 0.027681, 0.015224, 0.008373, 0.004605, 0.002533
115 0.898529 , 0.865051 , 0.769257 , 0.624054 , 0.448639 , 0.265289 ,
116 0.0959167 , -0.0412598 , -0.134338 , -0.178986 , -0.178528 , -0.142609 ,
117 -0.0849304 , -0.0205078 , 0.0369568 , 0.0773926 , 0.0955200 , 0.0912781 ,
118 0.0689392 , 0.0357056 , 0.0 , -0.0305481 , -0.0504150 , -0.0570068 ,
119 -0.0508423 , -0.0350037 , -0.0141602 , 0.00665283, 0.0230713 , 0.0323486 ,
120 0.0335388 , 0.0275879 , 0.0167847 , 0.00411987, -0.00747681, -0.0156860 ,
121 -0.0193481 , -0.0183716 , -0.0137634 , -0.00704956, 0.0 , 0.00582886 ,
122 0.00939941, 0.0103760 , 0.00903320, 0.00604248, 0.00238037, -0.00109863 ,
123 -0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
124 0.00103760, 0.00222778, 0.00277710, 0.00271606, 0.00213623, 0.00115967 ,
137 int mask = (1 << bits) - 1;
140 for(i=0; i<pulse_count; i++)
142 fc_v[i + tab1[pulse_indexes &
mask]] +=
143 (pulse_signs & 1) ? 8191 : -8192;
145 pulse_indexes >>= bits;
149 fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
155 int half_pulse_count,
int bits)
158 int mask = (1 << bits) - 1;
161 fixed_sparse->
n = 2 * half_pulse_count;
162 for (i = 0; i < half_pulse_count; i++) {
163 const int pos1 = gray_decode[fixed_index[2*i+1] &
mask] + i;
164 const int pos2 = gray_decode[fixed_index[2*i ] &
mask] + i;
165 const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
166 fixed_sparse->
x[2*i+1] = pos1;
167 fixed_sparse->
x[2*i ] = pos2;
168 fixed_sparse->
y[2*i+1] = sign;
169 fixed_sparse->
y[2*i ] = pos2 < pos1 ? -sign : sign;
177 int16_t weight_coeff_a,
178 int16_t weight_coeff_b,
187 out[i] = av_clip_int16((
188 in_a[i] * weight_coeff_a +
189 in_b[i] * weight_coeff_b +
194 float weight_coeff_a,
float weight_coeff_b,
int length)
199 out[i] = weight_coeff_a * in_a[i]
200 + weight_coeff_b * in_b[i];
208 float gain_scale_factor = 1.0;
209 float mem = *gain_mem;
211 if (postfilter_energ)
212 gain_scale_factor = sqrt(speech_energ / postfilter_energ);
214 gain_scale_factor *= 1.0 -
alpha;
216 for (i = 0; i <
size; i++) {
217 mem = alpha * mem + gain_scale_factor;
218 out[i] = in[i] *
mem;
225 float sum_of_squares,
const int n)
230 scalefactor = sqrt(sum_of_squares / scalefactor);
231 for (i = 0; i <
n; i++)
232 out[i] = in[i] * scalefactor;
239 for (i=0; i < in->
n; i++) {
241 float y = in->
y[i] * scale;
249 }
while (x < size && repeats);
257 for (i=0; i < in->
n; i++) {
264 }
while (x < size && repeats);
void ff_acelp_fc_pulse_per_track(int16_t *fc_v, const uint8_t *tab1, const uint8_t *tab2, int pulse_indexes, int pulse_signs, int pulse_count, int bits)
Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
static int shift(int a, int b)
static float alpha(float a)
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...
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.
void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
Clear array values set by set_fixed_vector.
const uint8_t ff_fc_2pulses_9bits_track1_gray[16]
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
Add fixed vector to an array from a sparse representation.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
void ff_acelp_vectors_init_mips(ACELPVContext *c)
const uint8_t ff_fc_4pulses_8bits_track_4[32]
Track|Pulse| Positions 4 | 3 | 3, 8, 13, 18, 23, 28, 33, 38, 43, 48, 53, 58, 63, 68, 73, 78 | | 4, 9, 14, 19, 24, 29, 34, 39, 44, 49, 54, 59, 64, 69, 74, 79
Sparse representation for the algebraic codebook (fixed) vector.
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)
void(* 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.
static const uint16_t mask[17]
const float ff_pow_0_7[10]
Table of pow(0.7,n)
simple assert() macros that are a bit more flexible than ISO C assert().
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)
void ff_acelp_weighted_vector_sum(int16_t *out, const int16_t *in_a, const int16_t *in_b, int16_t weight_coeff_a, int16_t weight_coeff_b, int16_t rounder, int shift, int length)
weighted sum of two vectors with rounding.
const uint8_t ff_fc_2pulses_9bits_track1[16]
Track|Pulse| Positions 1 | 0 | 1, 6, 11, 16, 21, 26, 31, 36 | | 3, 8, 13, 18, 23, 28...
Libavcodec external API header.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
const uint8_t ff_fc_4pulses_8bits_tracks_13[16]
Track|Pulse| Positions 1 | 0 | 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75 2 | 1 | 1, 6, 11, 16, 21, 26, 31, 36, 41, 46, 51, 56, 61, 66, 71, 76 3 | 2 | 2, 7, 12, 17, 22, 27, 32, 37, 42, 47, 52, 57, 62, 67, 72, 77
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
void ff_acelp_vectors_init(ACELPVContext *c)
Initialize ACELPVContext.
const uint8_t ff_fc_2pulses_9bits_track2_gray[32]
Track|Pulse| Positions 2 | 1 | 0, 7, 14, 20, 27, 34, 1, 21 | | 2, 9, 15, 22, 29, 35, 6, 26 | | 4,10, 17, 24, 30, 37, 11, 31 | | 5,12, 19, 25, 32, 39, 16, 36
static const uint8_t gray_decode[8]
3-bit Gray code to binary lookup table
common internal and external API header
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
const float ff_pow_0_55[10]
Table of pow(0.55,n)