Mercurial > libavcodec.hg
diff aac.c @ 7632:ac583bd8f8b3 libavcodec
Last hunk of the AAC decoder code to be OKed and build system and documentation
alterations as appropriate
author | superdump |
---|---|
date | Thu, 21 Aug 2008 07:21:26 +0000 |
parents | 3f6fc60f1ed3 |
children | bf34cb99da5b |
line wrap: on
line diff
--- a/aac.c Wed Aug 20 23:11:26 2008 +0000 +++ b/aac.c Thu Aug 21 07:21:26 2008 +0000 @@ -1104,6 +1104,42 @@ return res; } +/** + * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. + * + * @param decode 1 if tool is used normally, 0 if tool is used in LTP. + * @param coef spectral coefficients + */ +static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { + const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); + int w, filt, m, i, ib; + int bottom, top, order, start, end, size, inc; + float lpc[TNS_MAX_ORDER]; + + for (w = 0; w < ics->num_windows; w++) { + bottom = ics->num_swb; + for (filt = 0; filt < tns->n_filt[w]; filt++) { + top = bottom; + bottom = FFMAX(0, top - tns->length[w][filt]); + order = tns->order[w][filt]; + if (order == 0) + continue; + + /* tns_decode_coef + * FIXME: This duplicates the functionality of some double code in lpc.c. + */ + for (m = 0; m < order; m++) { + float tmp; + lpc[m] = tns->coef[w][filt][m]; + for (i = 0; i < m/2; i++) { + tmp = lpc[i]; + lpc[i] += lpc[m] * lpc[m-1-i]; + lpc[m-1-i] += lpc[m] * tmp; + } + if(m & 1) + lpc[i] += lpc[m] * lpc[i]; + } + start = ics->swb_offset[FFMIN(bottom, mmm)]; end = ics->swb_offset[FFMIN( top, mmm)]; if ((size = end - start) <= 0) @@ -1118,7 +1154,7 @@ // ar filter for (m = 0; m < size; m++, start += inc) for (i = 1; i <= FFMIN(m, order); i++) - coef[start] -= coef[start - i*inc] * lpc[i]; + coef[start] -= coef[start - i*inc] * lpc[i-1]; } } }