Go to the documentation of this file.
43 #define AMR_USE_16BIT_TABLES
128 ctx->first_frame = 1;
133 for (
i = 0;
i < 4;
i++)
157 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
158 ctx->fr_quality = (buf[0] & 0x4) == 0x4;
173 for (
i = 0;
i < 9;
i++)
176 for (
i = 0;
i < 7;
i++)
177 isf_q[
i + 9] =
dico2_isf[ind[1]][
i] * (1.0
f / (1 << 15));
179 for (
i = 0;
i < 5;
i++)
182 for (
i = 0;
i < 4;
i++)
185 for (
i = 0;
i < 7;
i++)
199 for (
i = 0;
i < 9;
i++)
202 for (
i = 0;
i < 7;
i++)
203 isf_q[
i + 9] =
dico2_isf[ind[1]][
i] * (1.0
f / (1 << 15));
205 for (
i = 0;
i < 3;
i++)
208 for (
i = 0;
i < 3;
i++)
211 for (
i = 0;
i < 3;
i++)
214 for (
i = 0;
i < 3;
i++)
217 for (
i = 0;
i < 4;
i++)
252 for (k = 0; k < 3; k++) {
255 isp_q[k][
i] = (1.0 -
c) * isp4_past[
i] +
c * isp_q[3][
i];
271 uint8_t *base_lag_int,
int subframe)
273 if (subframe == 0 || subframe == 2) {
274 if (pitch_index < 376) {
275 *lag_int = (pitch_index + 137) >> 2;
276 *lag_frac = pitch_index - (*lag_int << 2) + 136;
277 }
else if (pitch_index < 440) {
278 *lag_int = (pitch_index + 257 - 376) >> 1;
279 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
282 *lag_int = pitch_index - 280;
286 *base_lag_int =
av_clip(*lag_int - 8 - (*lag_frac < 0),
292 *lag_int = (pitch_index + 1) >> 2;
293 *lag_frac = pitch_index - (*lag_int << 2);
294 *lag_int += *base_lag_int;
304 uint8_t *base_lag_int,
int subframe,
enum Mode mode)
307 if (pitch_index < 116) {
308 *lag_int = (pitch_index + 69) >> 1;
309 *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
311 *lag_int = pitch_index - 24;
315 *base_lag_int =
av_clip(*lag_int - 8 - (*lag_frac < 0),
318 *lag_int = (pitch_index + 1) >> 1;
319 *lag_frac = (pitch_index - (*lag_int << 1)) * 2;
320 *lag_int += *base_lag_int;
336 int pitch_lag_int, pitch_lag_frac;
338 float *exc =
ctx->excitation;
343 &
ctx->base_pitch_lag, subframe,
mode);
346 &
ctx->base_pitch_lag, subframe);
348 ctx->pitch_lag_int = pitch_lag_int;
349 pitch_lag_int += pitch_lag_frac > 0;
353 ctx->acelpf_ctx.acelp_interpolatef(exc,
354 exc + 1 - pitch_lag_int,
356 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
361 if (amr_subframe->
ltp) {
365 ctx->pitch_vector[
i] = 0.18 * exc[
i - 1] + 0.64 * exc[
i] +
372 #define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len))
375 #define BIT_POS(x, p) (((x) >> (p)) & 1)
412 m - 1, off + half_2p);
418 int half_4p, subhalf_2p;
419 int b_offset = 1 << (m - 1);
427 m - 2, off + half_4p + subhalf_2p);
429 m - 1, off + half_4p);
435 m - 1, off + b_offset);
441 m - 1, off + b_offset);
447 m - 1, off + b_offset);
457 m - 1, off + half_3p);
464 int b_offset = 1 << (m - 1);
467 int half_other = b_offset - half_more;
472 m - 1, off + half_more);
474 m - 1, off + half_more);
478 m - 1, off + half_other);
480 m - 1, off + half_more);
484 m - 1, off + half_other);
486 m - 1, off + half_more);
492 m - 1, off + b_offset);
507 const uint16_t *pulse_lo,
const enum Mode mode)
517 for (
i = 0;
i < 2;
i++)
521 for (
i = 0;
i < 4;
i++)
525 for (
i = 0;
i < 4;
i++)
529 for (
i = 0;
i < 2;
i++)
531 for (
i = 2;
i < 4;
i++)
535 for (
i = 0;
i < 4;
i++)
539 for (
i = 0;
i < 4;
i++)
541 ((
int) pulse_hi[
i] << 14), 4, 1);
544 for (
i = 0;
i < 2;
i++)
546 ((
int) pulse_hi[
i] << 10), 4, 1);
547 for (
i = 2;
i < 4;
i++)
549 ((
int) pulse_hi[
i] << 14), 4, 1);
553 for (
i = 0;
i < 4;
i++)
555 ((
int) pulse_hi[
i] << 11), 4, 1);
561 for (
i = 0;
i < 4;
i++)
563 int pos = (
FFABS(sig_pos[
i][j]) - 1) * spacing +
i;
565 fixed_vector[
pos] += sig_pos[
i][j] < 0 ? -1.0 : 1.0;
578 float *fixed_gain_factor,
float *pitch_gain)
583 *pitch_gain = gains[0] * (1.0f / (1 << 14));
584 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
601 fixed_vector[
i] -= fixed_vector[
i - 1] *
ctx->tilt_coef;
605 fixed_vector[
i] += fixed_vector[
i -
ctx->pitch_lag_int] * 0.85;
618 float *f_vector,
float f_gain,
621 double p_ener = (
double)
ctx->dot_productf(p_vector, p_vector,
624 double f_ener = (
double)
ctx->dot_productf(f_vector, f_vector,
628 return (p_ener - f_ener) / (p_ener + f_ener + 0.01);
642 float *fixed_vector,
float *buf)
649 if (
ctx->pitch_gain[0] < 0.6) {
651 }
else if (
ctx->pitch_gain[0] < 0.9) {
657 if (
ctx->fixed_gain[0] > 3.0 *
ctx->fixed_gain[1]) {
658 if (ir_filter_nr < 2)
663 for (
i = 0;
i < 6;
i++)
664 if (
ctx->pitch_gain[
i] < 0.6)
670 if (ir_filter_nr >
ctx->prev_ir_filter_nr + 1)
675 ctx->prev_ir_filter_nr = ir_filter_nr;
679 if (ir_filter_nr < 2) {
711 acc += (isf[
i] - isf_past[
i]) * (isf[
i] - isf_past[
i]);
715 return FFMAX(0.0, 1.25 -
acc * 0.8 * 512);
730 float voice_fac,
float stab_fac)
732 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
738 if (fixed_gain < *prev_tr_gain) {
739 g0 =
FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
740 (6226 * (1.0
f / (1 << 15))));
742 g0 =
FFMAX(*prev_tr_gain, fixed_gain *
743 (27536 * (1.0
f / (1 << 15))));
747 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
759 float cpe = 0.125 * (1 + voice_fac);
760 float last = fixed_vector[0];
762 fixed_vector[0] -= cpe * fixed_vector[1];
765 float cur = fixed_vector[
i];
767 fixed_vector[
i] -= cpe * (last + fixed_vector[
i + 1]);
785 float fixed_gain,
const float *fixed_vector,
788 ctx->acelpv_ctx.weighted_vector_sumf(excitation,
ctx->pitch_vector, fixed_vector,
794 float energy =
ctx->celpm_ctx.dot_productf(excitation, excitation,
799 float pitch_factor = 0.25 *
ctx->pitch_gain[0] *
ctx->pitch_gain[0];
802 excitation[
i] += pitch_factor *
ctx->pitch_vector[
i];
808 ctx->celpf_ctx.celp_lp_synthesis_filterf(
samples, lpc, excitation,
825 out[0] = in[0] + m * mem[0];
846 int int_part = 0, frac_part;
849 for (j = 0; j < o_size / 5; j++) {
850 out[
i] = in[int_part];
854 for (k = 1; k < 5; k++) {
855 out[
i] =
ctx->dot_productf(in0 + int_part,
875 uint16_t hb_idx, uint8_t vad)
892 return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
905 const float *synth_exc,
float hb_gain)
908 float energy =
ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
916 energy * hb_gain * hb_gain,
929 float prod = (diff_isf[
i] -
mean) * (diff_isf[
i - lag] -
mean);
944 float diff_isf[
LP_ORDER - 2], diff_mean;
947 int i, j, i_max_corr;
953 diff_isf[
i] = isf[
i + 1] - isf[
i];
957 diff_mean += diff_isf[
i] * (1.0
f / (
LP_ORDER - 4));
961 for (
i = 0;
i < 3;
i++) {
964 if (corr_lag[
i] > corr_lag[i_max_corr])
970 isf[
i] = isf[
i - 1] + isf[
i - 1 - i_max_corr]
971 - isf[
i - 2 - i_max_corr];
974 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
979 diff_isf[j] =
scale * (isf[
i] - isf[
i - 1]);
983 if (diff_isf[
i] + diff_isf[
i - 1] < 5.0) {
984 if (diff_isf[
i] > diff_isf[
i - 1]) {
985 diff_isf[
i - 1] = 5.0 - diff_isf[
i];
987 diff_isf[
i] = 5.0 - diff_isf[
i - 1];
991 isf[
i] = isf[
i - 1] + diff_isf[j] * (1.0
f / (1 << 15));
1013 out[
i] = lpc[
i] * fac;
1030 const float *exc,
const float *isf,
const float *isf_past)
1039 ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf,
isfp_inter[subframe],
1069 #ifndef hb_fir_filter
1097 memmove(&
ctx->pitch_gain[1], &
ctx->pitch_gain[0], 5 *
sizeof(
float));
1098 memmove(&
ctx->fixed_gain[1], &
ctx->fixed_gain[0], 1 *
sizeof(
float));
1109 int *got_frame_ptr,
AVPacket *avpkt)
1112 const uint8_t *buf = avpkt->
data;
1113 int buf_size = avpkt->
size;
1124 int expected_fr_size, header_size;
1126 float fixed_gain_factor;
1127 float *synth_fixed_vector;
1128 float synth_fixed_gain;
1129 float voice_fac, stab_fac;
1134 float *buf_out = (
float *)
frame->extended_data[ch];
1137 expected_fr_size = ((
cf_sizes_wb[
ctx->fr_cur_mode] + 7) >> 3) + 1;
1139 if (!
ctx->fr_quality)
1146 buf += expected_fr_size;
1147 buf_size -= expected_fr_size;
1152 "Invalid mode %d\n",
ctx->fr_cur_mode);
1156 if (buf_size < expected_fr_size) {
1158 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1187 if (
ctx->first_frame) {
1188 ctx->first_frame = 0;
1189 memcpy(
ctx->isp_sub4_past,
ctx->isp[3],
LP_ORDER *
sizeof(
double));
1204 cur_subframe->
pul_il,
ctx->fr_cur_mode);
1209 &fixed_gain_factor, &
ctx->pitch_gain[0]);
1211 ctx->fixed_gain[0] =
1213 ctx->celpm_ctx.dot_productf(
ctx->fixed_vector,
1217 ctx->prediction_error,
1222 ctx->fixed_vector,
ctx->fixed_gain[0],
1224 ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1228 ctx->excitation[
i] *=
ctx->pitch_gain[0];
1229 ctx->excitation[
i] +=
ctx->fixed_gain[0] *
ctx->fixed_vector[
i];
1235 voice_fac, stab_fac);
1249 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&
ctx->samples_up[
UPS_MEM_SIZE],
1257 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
1267 hb_exc,
ctx->isf_cur,
ctx->isf_past_final);
1279 sub_buf[
i] = (sub_buf[
i] + hb_samples[
i]) * (1.0f / (1 << 15));
1287 memcpy(
ctx->isf_past_final,
ctx->isf_cur,
LP_ORDER *
sizeof(
float));
1289 buf += expected_fr_size;
1290 buf_size -= expected_fr_size;
1295 return buf - avpkt->
data;
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)
@ AV_SAMPLE_FMT_FLTP
float, planar
static void pitch_enhancer(float *fixed_vector, float voice_fac)
Filter the fixed_vector to emphasize the higher frequencies.
uint8_t base_pitch_lag
integer part of pitch lag for the next relative subframe
#define UPS_FIR_SIZE
upsampling filter size
float demph_mem[1]
previous value in the de-emphasis filter
uint8_t prev_ir_filter_nr
previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
double isp_sub4_past[LP_ORDER]
ISP vector for the 4th subframe of the previous 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 void decode_2p_track(int *out, int code, int m, int off)
code: 2m+1 bits
static const int16_t dico22_isf[128][3]
static const float upsample_fir[4][24]
Interpolation coefficients for 5/4 signal upsampling Table from the reference source was reordered fo...
static const uint16_t qua_hb_gain[16]
High band quantized gains for 23k85 in Q14.
int sample_rate
samples per second
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
float samples_az[LP_ORDER+AMRWB_SFR_SIZE]
low-band samples and memory from synthesis at 12.8kHz
static float sub(float src0, float src1)
float prev_sparse_fixed_gain
previous fixed gain; used by anti-sparseness to determine "onset"
void ff_amrwb_lsp2lpc(const double *lsp, float *lp, int lp_order)
LSP to LP conversion (5.2.4 of AMR-WB)
static const int16_t dico2_isf[256][7]
uint16_t hb_gain
high-band energy index (mode 23k85 only)
static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
This structure describes decoded (raw) audio or video data.
uint16_t adap
adaptive codebook index
void ff_acelp_filter_init(ACELPFContext *c)
Initialize ACELPFContext.
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).
int nb_channels
Number of channels in this layout.
float pitch_gain[6]
quantified pitch gains for the current and previous five subframes
static int amrwb_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
CELPMContext celpm_ctx
context for fixed point math operations
#define AMRWB_P_DELAY_MAX
maximum pitch delay value
uint8_t pitch_lag_int
integer part of pitch lag of the previous subframe
static const float bpf_6_7_coef[31]
High-band post-processing FIR filters coefficients from Q15.
static const int16_t dico23_isf[128][3]
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
uint16_t isp_id[7]
index of ISP subvectors
uint8_t fr_quality
frame quality index (FQI)
static av_always_inline float scale(float x, float s)
float * excitation
points to current excitation in excitation_buf[]
float samples_hb[LP_ORDER_16k+AMRWB_SFR_SIZE_16k]
high-band samples and memory from synthesis at 16kHz
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 update_sub_state(AMRWBContext *ctx)
Update context state before the next subframe.
#define BIT_STR(x, lsb, len)
Get x bits in the index interval [lsb,lsb+len-1] inclusive.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
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 subfr...
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 ...
static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
Decode the frame header in the "MIME/storage" format.
Mode
Frame type (Table 1a in 3GPP TS 26.101)
AMRWBFrame frame
AMRWB parameters decoded from bitstream.
static const int16_t dico21_isf_36b[128][5]
#define FF_CODEC_DECODE_CB(func)
float fixed_vector[AMRWB_SFR_SIZE]
algebraic codebook (fixed) vector for current subframe
#define BIT_POS(x, p)
Get the bit at specified position.
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
float lp_coef[4][LP_ORDER]
Linear Prediction Coefficients from ISP vector.
static const int16_t isf_mean[LP_ORDER]
Means of ISF vectors in Q15.
static const float energy_pred_fac[4]
4-tap moving average prediction coefficients in reverse order
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.
int(* init)(AVBSFContext *ctx)
static const float hpf_zeros[2]
High-pass filters coefficients for 31 Hz and 400 Hz cutoff.
#define MIN_ISF_SPACING
minimum isf gap
void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
Floating point version of ff_acelp_lsf2lsp()
uint16_t ltp
ltp-filtering flag
enum Mode fr_cur_mode
mode index of current frame
static const uint16_t *const amr_bit_orderings_by_mode[]
Reordering array addresses for each mode.
static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
Apply pitch sharpening filters to the fixed codebook vector.
#define CODEC_LONG_NAME(str)
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 gai...
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
float samples_up[UPS_MEM_SIZE+AMRWB_SFR_SIZE]
low-band samples and memory processed for upsampling
@ MODE_SID
comfort noise frame
uint16_t vq_gain
VQ adaptive and innovative gains.
static const int16_t qua_gain_7b[128][2]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static const uint16_t cf_sizes_wb[]
Core frame sizes in each mode.
static const float *const ir_filters_lookup[2]
float lpf_7_mem[HB_FIR_SIZE]
previous values in the high-band low pass filter
static const float hpf_31_poles[2]
#define AMRWB_SFR_SIZE
samples per subframe at 12.8 kHz
static const int16_t dico21_isf[64][3]
#define AMRWB_SFR_SIZE_16k
samples per subframe at 16 kHz
static const int16_t dico25_isf[32][4]
static const int16_t dico23_isf_36b[64][7]
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 const uint8_t pulses_nb_per_mode_tr[][4]
[i][j] is the number of pulses present in track j at mode i
static float auto_correlation(float *diff_isf, float mean, int lag)
Calculate the auto-correlation for the ISF difference vector.
#define AMRWB_P_DELAY_MIN
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
AVLFG prng
random number generator for white noise excitation
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
void ff_acelp_vectors_init(ACELPVContext *c)
Initialize ACELPVContext.
Context structure for the Lagged Fibonacci PRNG.
void ff_celp_math_init(CELPMContext *c)
Initialize CELPMContext.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
uint8_t first_frame
flag active during decoding of the first frame
An AVChannelLayout holds information about the channel layout of audio data.
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...
#define ENERGY_MEAN
mean innovation energy (dB) in all modes
#define LP_ORDER_16k
lpc filter order at 16kHz
static const float hpf_31_gain
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)
enum AVSampleFormat sample_fmt
audio sample format
static void decode_5p_track(int *out, int code, int m, int off)
code: 5m bits
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.
float pitch_vector[AMRWB_SFR_SIZE]
adaptive codebook (pitch) vector for current subframe
float hpf_400_mem[2]
previous values in the high pass filters
static const int16_t dico22_isf_36b[128][4]
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
ACELPFContext acelpf_ctx
context for filters for ACELP-based codecs
static const int16_t dico24_isf[32][3]
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 const int16_t isf_init[LP_ORDER]
Initialization tables for the processed ISF vector in Q15.
static float stability_factor(const float *isf, const float *isf_past)
Calculate a stability factor {teta} based on distance between current and past isf.
uint16_t vad
voice activity detection flag
void ff_celp_filter_init(CELPFContext *c)
Initialize CELPFContext.
float isf_past_final[LP_ORDER]
final processed ISF vector of the previous frame
float fixed_gain[2]
quantified fixed gains for the current and previous subframes
static void isf_add_mean_and_past(float *isf_q, float *isf_past)
Apply mean and past ISF values using the prediction factor.
#define i(width, name, range_min, range_max)
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
AVSampleFormat
Audio sample formats.
float bpf_6_7_mem[HB_FIR_SIZE]
previous values in the high-band band pass filter
const char * name
Name of the codec implementation.
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
static void decode_4p_track(int *out, int code, int m, int off)
code: 4m bits
double isp[4][LP_ORDER]
ISP vectors from current frame.
static const float lpf_7_coef[31]
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
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 const float ac_inter[65]
Coefficients for FIR interpolation of excitation vector at pitch lag resulting the adaptive codebook ...
uint16_t pul_ih[4]
MSBs part of codebook index (high modes only)
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.
main external API structure.
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 pha...
static const float hpf_400_gain
static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
float isf_cur[LP_ORDER]
working ISF vector from current frame
Filter the word “frame” indicates either a video frame or a group of audio samples
static float mean(const float *input, int size)
static const float isfp_inter[4]
ISF/ISP interpolation coefficients for each subframe.
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...
#define LP_ORDER
linear predictive coding filter order
static void decode_3p_track(int *out, int code, int m, int off)
code: 3m+1 bits
float prev_tr_gain
previous initial gain used by noise enhancer for threshold
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 codebo...
#define avpriv_request_sample(...)
float prediction_error[4]
quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
#define AV_CHANNEL_LAYOUT_MONO
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) i...
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.
This structure stores compressed data.
float excitation_buf[AMRWB_P_DELAY_MAX+LP_ORDER+2+AMRWB_SFR_SIZE]
current excitation and all necessary excitation history
#define HB_FIR_SIZE
amount of past data needed by HB filters
static av_cold int amrwb_decode_init(AVCodecContext *avctx)
static const int16_t dico1_isf[256][9]
Indexed tables for retrieval of quantized ISF vectors in Q15.
static void decode_6p_track(int *out, int code, int m, int off)
code: 6m-2 bits
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.
AMRWBSubFrame subframe[4]
data for subframes
float isf_q_past[LP_ORDER]
quantized ISF vector of the previous frame
const FFCodec ff_amrwb_decoder
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
#define MIN_ENERGY
Initial energy in dB.
static const int16_t qua_gain_6b[64][2]
Tables for decoding quantized gains { pitch (Q14), fixed factor (Q11) }.
uint16_t pul_il[4]
LSBs part of codebook index.
static __device__ float truncf(float a)
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.
ACELPVContext acelpv_ctx
context for vector operations for ACELP-based codecs
float tilt_coef
{beta_1} related to the voicing of the previous subframe
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 t...
CELPFContext celpf_ctx
context for filters for CELP-based codecs
static const float hpf_400_poles[2]
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.
#define PREEMPH_FAC
factor used to de-emphasize synthesis