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
26 * @remark FFmpeg merging spearheaded by Kenan Gillet
32 #include "bitstream.h"
35 #include "qcelpdata.h"
37 #include "celp_math.h"
38 #include "celp_filters.h"
43 static void weighted_vector_sumf(float *out, const float *in_a,
44 const float *in_b, float weight_coeff_a,
45 float weight_coeff_b, int length)
49 for(i=0; i<length; i++)
50 out[i] = weight_coeff_a * in_a[i]
51 + weight_coeff_b * in_b[i];
55 * Initialize the speech codec according to the specification.
57 * TIA/EIA/IS-733 2.4.9
59 static av_cold int qcelp_decode_init(AVCodecContext *avctx) {
60 QCELPContext *q = avctx->priv_data;
63 avctx->sample_fmt = SAMPLE_FMT_FLT;
65 for (i = 0; i < 10; i++)
66 q->prev_lspf[i] = (i + 1) / 11.;
72 * Computes the scaled codebook vector Cdn From INDEX and GAIN
75 * The specification lacks some information here.
77 * TIA/EIA/IS-733 has an omission on the codebook index determination
78 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
79 * you have to subtract the decoded index parameter from the given scaled
80 * codebook vector index 'n' to get the desired circular codebook index, but
81 * it does not mention that you have to clamp 'n' to [0-9] in order to get
82 * RI-compliant results.
84 * The reason for this mistake seems to be the fact they forgot to mention you
85 * have to do these calculations per codebook subframe and adjust given
86 * equation values accordingly.
88 * @param q the context
89 * @param gain array holding the 4 pitch subframe gain values
90 * @param cdn_vector array for the generated scaled codebook vector
92 static void compute_svector(const QCELPContext *q,
96 uint16_t cbseed, cindex;
97 float *rnd, tmp_gain, fir_filter_value;
99 switch (q->framerate) {
101 for (i = 0; i < 16; i++) {
102 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
103 cindex = -q->cindex[i];
104 for (j = 0; j < 10; j++)
105 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
109 for (i = 0; i < 4; i++) {
110 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
111 cindex = -q->cindex[i];
112 for (j = 0; j < 40; j++)
113 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
117 cbseed = (0x0003 & q->lspv[4])<<14 |
118 (0x003F & q->lspv[3])<< 8 |
119 (0x0060 & q->lspv[2])<< 1 |
120 (0x0007 & q->lspv[1])<< 3 |
121 (0x0038 & q->lspv[0])>> 3 ;
122 rnd = q->rnd_fir_filter_mem + 20;
123 for (i = 0; i < 8; i++) {
124 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
125 for (k = 0; k < 20; k++) {
126 cbseed = 521 * cbseed + 259;
127 *rnd = (int16_t)cbseed;
130 fir_filter_value = 0.0;
131 for (j = 0; j < 10; j++)
132 fir_filter_value += qcelp_rnd_fir_coefs[j ] * (rnd[-j ] + rnd[-20+j]);
133 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
135 *cdn_vector++ = tmp_gain * fir_filter_value;
139 memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
142 cbseed = q->first16bits;
143 for (i = 0; i < 8; i++) {
144 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
145 for (j = 0; j < 20; j++) {
146 cbseed = 521 * cbseed + 259;
147 *cdn_vector++ = tmp_gain * (int16_t)cbseed;
152 cbseed = -44; // random codebook index
153 for (i = 0; i < 4; i++) {
154 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
155 for (j = 0; j < 40; j++)
156 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
163 * Apply generic gain control.
165 * @param v_out output vector
166 * @param v_in gain-controlled vector
167 * @param v_ref vector to control gain of
169 * FIXME: If v_ref is a zero vector, it energy is zero
170 * and the behavior of the gain control is
171 * undefined in the specs.
173 * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
175 static void apply_gain_ctrl(float *v_out,
181 for (i = 0, j = 0; i < 4; i++) {
182 scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
184 scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40) / scalefactor);
186 av_log_missing_feature(NULL, "Zero energy for gain control", 1);
187 for (len = j + 40; j < len; j++)
188 v_out[j] = scalefactor * v_in[j];
193 * Apply filter in pitch-subframe steps.
195 * @param memory buffer for the previous state of the filter
196 * - must be able to contain 303 elements
197 * - the 143 first elements are from the previous state
198 * - the next 160 are for output
199 * @param v_in input filter vector
200 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
201 * @param lag per-subframe lag array, each element is
202 * - between 16 and 143 if its corresponding pfrac is 0,
203 * - between 16 and 139 otherwise
204 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 otherwise
206 * @return filter output vector
208 static const float *do_pitchfilter(float memory[303], const float v_in[160],
209 const float gain[4], const uint8_t *lag,
210 const uint8_t pfrac[4])
213 float *v_lag, *v_out;
216 v_out = memory + 143; // Output vector starts at memory[143].
222 v_lag = memory + 143 + 40 * i - lag[i];
223 for(v_len=v_in+40; v_in<v_len; v_in++)
225 if(pfrac[i]) // If it is a fractional lag...
227 for(j=0, *v_out=0.; j<4; j++)
228 *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
232 *v_out = *v_in + gain[i] * *v_out;
239 memcpy(v_out, v_in, 40 * sizeof(float));
245 memmove(memory, memory + 160, 143 * sizeof(float));
250 * Interpolates LSP frequencies and computes LPC coefficients
251 * for a given framerate & pitch subframe.
253 * TIA/EIA/IS-733 2.4.3.3.4
255 * @param q the context
256 * @param curr_lspf LSP frequencies vector of the current frame
257 * @param lpc float vector for the resulting LPC
258 * @param subframe_num frame number in decoded stream
260 void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
261 const int subframe_num)
263 float interpolated_lspf[10];
266 if(q->framerate >= RATE_QUARTER)
267 weight = 0.25 * (subframe_num + 1);
268 else if(q->framerate == RATE_OCTAVE && !subframe_num)
275 weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
276 weight, 1.0 - weight, 10);
277 qcelp_lspf2lpc(interpolated_lspf, lpc);
278 }else if(q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num))
279 qcelp_lspf2lpc(curr_lspf, lpc);
282 static int buf_size2framerate(const int buf_size)
301 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
304 av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
308 AVCodec qcelp_decoder =
311 .type = CODEC_TYPE_AUDIO,
312 .id = CODEC_ID_QCELP,
313 .init = qcelp_decode_init,
314 .decode = qcelp_decode_frame,
315 .priv_data_size = sizeof(QCELPContext),
316 .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),