3 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 * @author Reynaldo H. Verdejo Pinochet
31 #include "bitstream.h"
34 #include "qcelpdata.h"
36 #include "celp_math.h"
37 #include "celp_filters.h"
42 static void weighted_vector_sumf(float *out, const float *in_a,
43 const float *in_b, float weight_coeff_a,
44 float weight_coeff_b, int length)
48 for(i=0; i<length; i++)
49 out[i] = weight_coeff_a * in_a[i]
50 + weight_coeff_b * in_b[i];
54 * Initialize the speech codec according to the specification.
56 * TIA/EIA/IS-733 2.4.9
58 static av_cold int qcelp_decode_init(AVCodecContext *avctx) {
59 QCELPContext *q = avctx->priv_data;
62 avctx->sample_fmt = SAMPLE_FMT_FLT;
64 for (i = 0; i < 10; i++)
65 q->prev_lspf[i] = (i + 1) / 11.;
71 * Computes the scaled codebook vector Cdn From INDEX and GAIN
74 * The specification lacks some information here.
76 * TIA/EIA/IS-733 has an omission on the codebook index determination
77 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
78 * you have to subtract the decoded index parameter from the given scaled
79 * codebook vector index 'n' to get the desired circular codebook index, but
80 * it does not mention that you have to clamp 'n' to [0-9] in order to get
81 * RI-compliant results.
83 * The reason for this mistake seems to be the fact they forgot to mention you
84 * have to do these calculations per codebook subframe and adjust given
85 * equation values accordingly.
87 * @param q the context
88 * @param gain array holding the 4 pitch subframe gain values
89 * @param cdn_vector array for the generated scaled codebook vector
91 static void compute_svector(const QCELPContext *q,
95 uint16_t cbseed, cindex;
96 float *rnd, tmp_gain, fir_filter_value;
98 switch (q->framerate) {
100 for (i = 0; i < 16; i++) {
101 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
102 cindex = -q->cindex[i];
103 for (j = 0; j < 10; j++)
104 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
108 for (i = 0; i < 4; i++) {
109 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
110 cindex = -q->cindex[i];
111 for (j = 0; j < 40; j++)
112 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
116 cbseed = (0x0003 & q->lspv[4])<<14 |
117 (0x003F & q->lspv[3])<< 8 |
118 (0x0060 & q->lspv[2])<< 1 |
119 (0x0007 & q->lspv[1])<< 3 |
120 (0x0038 & q->lspv[0])>> 3 ;
121 rnd = q->rnd_fir_filter_mem + 20;
122 for (i = 0; i < 8; i++) {
123 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
124 for (k = 0; k < 20; k++) {
125 cbseed = 521 * cbseed + 259;
126 *rnd = (int16_t)cbseed;
129 fir_filter_value = 0.0;
130 for (j = 0; j < 10; j++)
131 fir_filter_value += qcelp_rnd_fir_coefs[j ] * (rnd[-j ] + rnd[-20+j]);
132 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
134 *cdn_vector++ = tmp_gain * fir_filter_value;
138 memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
141 cbseed = q->first16bits;
142 for (i = 0; i < 8; i++) {
143 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
144 for (j = 0; j < 20; j++) {
145 cbseed = 521 * cbseed + 259;
146 *cdn_vector++ = tmp_gain * (int16_t)cbseed;
151 cbseed = -44; // random codebook index
152 for (i = 0; i < 4; i++) {
153 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
154 for (j = 0; j < 40; j++)
155 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
162 * Apply generic gain control.
164 * @param v_out output vector
165 * @param v_in gain-controlled vector
166 * @param v_ref vector to control gain of
168 * FIXME: If v_ref is a zero vector, it energy is zero
169 * and the behavior of the gain control is
170 * undefined in the specs.
172 * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
174 static void apply_gain_ctrl(float *v_out,
180 for (i = 0, j = 0; i < 4; i++) {
181 scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
183 scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40) / scalefactor);
185 av_log_missing_feature(NULL, "Zero energy for gain control", 1);
186 for (len = j + 40; j < len; j++)
187 v_out[j] = scalefactor * v_in[j];
192 * Apply filter in pitch-subframe steps.
194 * @param memory buffer for the previous state of the filter
195 * - must be able to contain 303 elements
196 * - the 143 first elements are from the previous state
197 * - the next 160 are for output
198 * @param v_in input filter vector
199 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
200 * @param lag per-subframe lag array, each element is
201 * - between 16 and 143 if its corresponding pfrac is 0,
202 * - between 16 and 139 otherwise
203 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 otherwise
205 * @return filter output vector
207 static const float *do_pitchfilter(float memory[303], const float v_in[160],
208 const float gain[4], const uint8_t *lag,
209 const uint8_t pfrac[4])
212 float *v_lag, *v_out;
215 v_out = memory + 143; // Output vector starts at memory[143].
221 v_lag = memory + 143 + 40 * i - lag[i];
222 for(v_len=v_in+40; v_in<v_len; v_in++)
224 if(pfrac[i]) // If it is a fractional lag...
226 for(j=0, *v_out=0.; j<4; j++)
227 *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
231 *v_out = *v_in + gain[i] * *v_out;
238 memcpy(v_out, v_in, 40 * sizeof(float));
244 memmove(memory, memory + 160, 143 * sizeof(float));
249 * Interpolates LSP frequencies and computes LPC coefficients
250 * for a given framerate & pitch subframe.
252 * TIA/EIA/IS-733 2.4.3.3.4
254 * @param q the context
255 * @param curr_lspf LSP frequencies vector of the current frame
256 * @param lpc float vector for the resulting LPC
257 * @param subframe_num frame number in decoded stream
259 void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
260 const int subframe_num)
262 float interpolated_lspf[10];
265 if(q->framerate >= RATE_QUARTER)
266 weight = 0.25 * (subframe_num + 1);
267 else if(q->framerate == RATE_OCTAVE && !subframe_num)
274 weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
275 weight, 1.0 - weight, 10);
276 qcelp_lspf2lpc(interpolated_lspf, lpc);
277 }else if(q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num))
278 qcelp_lspf2lpc(curr_lspf, lpc);
281 static int buf_size2framerate(const int buf_size)
300 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
303 av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
307 AVCodec qcelp_decoder =
310 .type = CODEC_TYPE_AUDIO,
311 .id = CODEC_ID_QCELP,
312 .init = qcelp_decode_init,
313 .decode = qcelp_decode_frame,
314 .priv_data_size = sizeof(QCELPContext),
315 .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),