Mercurial > libavcodec.hg
view qcelpdec.c @ 8158:88a1babf9bde libavcodec
cosmetics: Group ARM objects sections together.
author | diego |
---|---|
date | Mon, 17 Nov 2008 15:36:28 +0000 |
parents | e2c068cb210a |
children | cc1e8c59f1e8 |
line wrap: on
line source
/* * QCELP decoder * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file qcelpdec.c * QCELP decoder * @author Reynaldo H. Verdejo Pinochet * @remark FFmpeg merging spearheaded by Kenan Gillet */ #include <stddef.h> #include "avcodec.h" #include "bitstream.h" #include "qcelp.h" #include "qcelpdata.h" #include "celp_math.h" #include "celp_filters.h" #undef NDEBUG #include <assert.h> static void weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length) { int i; for(i=0; i<length; i++) out[i] = weight_coeff_a * in_a[i] + weight_coeff_b * in_b[i]; } /** * Initialize the speech codec according to the specification. * * TIA/EIA/IS-733 2.4.9 */ static av_cold int qcelp_decode_init(AVCodecContext *avctx) { QCELPContext *q = avctx->priv_data; int i; avctx->sample_fmt = SAMPLE_FMT_FLT; for (i = 0; i < 10; i++) q->prev_lspf[i] = (i + 1) / 11.; return 0; } /** * Computes the scaled codebook vector Cdn From INDEX and GAIN * for all rates. * * The specification lacks some information here. * * TIA/EIA/IS-733 has an omission on the codebook index determination * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says * you have to subtract the decoded index parameter from the given scaled * codebook vector index 'n' to get the desired circular codebook index, but * it does not mention that you have to clamp 'n' to [0-9] in order to get * RI-compliant results. * * The reason for this mistake seems to be the fact they forgot to mention you * have to do these calculations per codebook subframe and adjust given * equation values accordingly. * * @param q the context * @param gain array holding the 4 pitch subframe gain values * @param cdn_vector array for the generated scaled codebook vector */ static void compute_svector(const QCELPContext *q, const float *gain, float *cdn_vector) { int i, j, k; uint16_t cbseed, cindex; float *rnd, tmp_gain, fir_filter_value; switch (q->framerate) { case RATE_FULL: for (i = 0; i < 16; i++) { tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO; cindex = -q->cindex[i]; for (j = 0; j < 10; j++) *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127]; } break; case RATE_HALF: for (i = 0; i < 4; i++) { tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO; cindex = -q->cindex[i]; for (j = 0; j < 40; j++) *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127]; } break; case RATE_QUARTER: cbseed = (0x0003 & q->lspv[4])<<14 | (0x003F & q->lspv[3])<< 8 | (0x0060 & q->lspv[2])<< 1 | (0x0007 & q->lspv[1])<< 3 | (0x0038 & q->lspv[0])>> 3 ; rnd = q->rnd_fir_filter_mem + 20; for (i = 0; i < 8; i++) { tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0); for (k = 0; k < 20; k++) { cbseed = 521 * cbseed + 259; *rnd = (int16_t)cbseed; // FIR filter fir_filter_value = 0.0; for (j = 0; j < 10; j++) fir_filter_value += qcelp_rnd_fir_coefs[j ] * (rnd[-j ] + rnd[-20+j]); fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10]; *cdn_vector++ = tmp_gain * fir_filter_value; rnd++; } } memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float)); break; case RATE_OCTAVE: cbseed = q->first16bits; for (i = 0; i < 8; i++) { tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0); for (j = 0; j < 20; j++) { cbseed = 521 * cbseed + 259; *cdn_vector++ = tmp_gain * (int16_t)cbseed; } } break; case I_F_Q: cbseed = -44; // random codebook index for (i = 0; i < 4; i++) { tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO; for (j = 0; j < 40; j++) *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127]; } break; } } /** * Apply generic gain control. * * @param v_out output vector * @param v_in gain-controlled vector * @param v_ref vector to control gain of * * FIXME: If v_ref is a zero vector, it energy is zero * and the behavior of the gain control is * undefined in the specs. * * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6 */ static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in) { int i, j, len; float scalefactor; for (i = 0, j = 0; i < 4; i++) { scalefactor = ff_dot_productf(v_in + j, v_in + j, 40); if (scalefactor) scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40) / scalefactor); else av_log_missing_feature(NULL, "Zero energy for gain control", 1); for (len = j + 40; j < len; j++) v_out[j] = scalefactor * v_in[j]; } } /** * Apply filter in pitch-subframe steps. * * @param memory buffer for the previous state of the filter * - must be able to contain 303 elements * - the 143 first elements are from the previous state * - the next 160 are for output * @param v_in input filter vector * @param gain per-subframe gain array, each element is between 0.0 and 2.0 * @param lag per-subframe lag array, each element is * - between 16 and 143 if its corresponding pfrac is 0, * - between 16 and 139 otherwise * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 otherwise * * @return filter output vector */ static const float *do_pitchfilter(float memory[303], const float v_in[160], const float gain[4], const uint8_t *lag, const uint8_t pfrac[4]) { int i, j; float *v_lag, *v_out; const float *v_len; v_out = memory + 143; // Output vector starts at memory[143]. for(i=0; i<4; i++) { if(gain[i]) { v_lag = memory + 143 + 40 * i - lag[i]; for(v_len=v_in+40; v_in<v_len; v_in++) { if(pfrac[i]) // If it is a fractional lag... { for(j=0, *v_out=0.; j<4; j++) *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]); }else *v_out = *v_lag; *v_out = *v_in + gain[i] * *v_out; v_lag++; v_out++; } }else { memcpy(v_out, v_in, 40 * sizeof(float)); v_in += 40; v_out += 40; } } memmove(memory, memory + 160, 143 * sizeof(float)); return memory + 143; } /** * Interpolates LSP frequencies and computes LPC coefficients * for a given framerate & pitch subframe. * * TIA/EIA/IS-733 2.4.3.3.4 * * @param q the context * @param curr_lspf LSP frequencies vector of the current frame * @param lpc float vector for the resulting LPC * @param subframe_num frame number in decoded stream */ void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc, const int subframe_num) { float interpolated_lspf[10]; float weight; if(q->framerate >= RATE_QUARTER) weight = 0.25 * (subframe_num + 1); else if(q->framerate == RATE_OCTAVE && !subframe_num) weight = 0.625; else weight = 1.0; if(weight != 1.0) { weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, weight, 1.0 - weight, 10); qcelp_lspf2lpc(interpolated_lspf, lpc); }else if(q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num)) qcelp_lspf2lpc(curr_lspf, lpc); } static int buf_size2framerate(const int buf_size) { switch(buf_size) { case 35: return RATE_FULL; case 17: return RATE_HALF; case 8: return RATE_QUARTER; case 4: return RATE_OCTAVE; case 1: return SILENCE; } return -1; } static void warn_insufficient_frame_quality(AVCodecContext *avctx, const char *message) { av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number, message); } AVCodec qcelp_decoder = { .name = "qcelp", .type = CODEC_TYPE_AUDIO, .id = CODEC_ID_QCELP, .init = qcelp_decode_init, .decode = qcelp_decode_frame, .priv_data_size = sizeof(QCELPContext), .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"), };