diff options
Diffstat (limited to 'libavcodec/ac3enc_template.c')
| -rw-r--r-- | libavcodec/ac3enc_template.c | 40 |
1 files changed, 15 insertions, 25 deletions
diff --git a/libavcodec/ac3enc_template.c b/libavcodec/ac3enc_template.c index ef40b5a..be65987 100644 --- a/libavcodec/ac3enc_template.c +++ b/libavcodec/ac3enc_template.c @@ -4,20 +4,20 @@ * Copyright (c) 2006-2011 Justin Ruggles <justin.ruggles@gmail.com> * Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de> * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -43,7 +43,7 @@ int AC3_NAME(allocate_sample_buffers)(AC3EncodeContext *s) FF_ALLOC_OR_GOTO(s->avctx, s->windowed_samples, AC3_WINDOW_SIZE * sizeof(*s->windowed_samples), alloc_fail); - FF_ALLOC_OR_GOTO(s->avctx, s->planar_samples, s->channels * sizeof(*s->planar_samples), + FF_ALLOC_ARRAY_OR_GOTO(s->avctx, s->planar_samples, s->channels, sizeof(*s->planar_samples), alloc_fail); for (ch = 0; ch < s->channels; ch++) { FF_ALLOCZ_OR_GOTO(s->avctx, s->planar_samples[ch], @@ -59,7 +59,7 @@ alloc_fail: /* * Copy input samples. - * Channels are reordered from Libav's default order to AC-3 order. + * Channels are reordered from FFmpeg's default order to AC-3 order. */ static void copy_input_samples(AC3EncodeContext *s, SampleType **samples) { @@ -94,7 +94,7 @@ static void apply_mdct(AC3EncodeContext *s) const SampleType *input_samples = &s->planar_samples[ch][blk * AC3_BLOCK_SIZE]; #if CONFIG_AC3ENC_FLOAT - s->fdsp.vector_fmul(s->windowed_samples, input_samples, + s->fdsp->vector_fmul(s->windowed_samples, input_samples, s->mdct_window, AC3_WINDOW_SIZE); #else s->ac3dsp.apply_window_int16(s->windowed_samples, input_samples, @@ -122,7 +122,7 @@ static void apply_channel_coupling(AC3EncodeContext *s) #else int32_t (*fixed_cpl_coords)[AC3_MAX_CHANNELS][16] = cpl_coords; #endif - int blk, ch, bnd, i, j; + int av_uninit(blk), ch, bnd, i, j; CoefSumType energy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][16] = {{{0}}}; int cpl_start, num_cpl_coefs; @@ -325,8 +325,8 @@ static void apply_channel_coupling(AC3EncodeContext *s) static void compute_rematrixing_strategy(AC3EncodeContext *s) { int nb_coefs; - int blk, bnd, i; - AC3Block *block, *block0; + int blk, bnd; + AC3Block *block, *block0 = NULL; if (s->channel_mode != AC3_CHMODE_STEREO) return; @@ -350,20 +350,12 @@ static void compute_rematrixing_strategy(AC3EncodeContext *s) } for (bnd = 0; bnd < block->num_rematrixing_bands; bnd++) { - /* calculate calculate sum of squared coeffs for one band in one block */ + /* calculate sum of squared coeffs for one band in one block */ int start = ff_ac3_rematrix_band_tab[bnd]; int end = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]); - CoefSumType sum[4] = {0,}; - for (i = start; i < end; i++) { - CoefType lt = block->mdct_coef[1][i]; - CoefType rt = block->mdct_coef[2][i]; - CoefType md = lt + rt; - CoefType sd = lt - rt; - MAC_COEF(sum[0], lt, lt); - MAC_COEF(sum[1], rt, rt); - MAC_COEF(sum[2], md, md); - MAC_COEF(sum[3], sd, sd); - } + CoefSumType sum[4]; + sum_square_butterfly(s, sum, block->mdct_coef[1] + start, + block->mdct_coef[2] + start, end - start); /* compare sums to determine if rematrixing will be used for this band */ if (FFMIN(sum[2], sum[3]) < FFMIN(sum[0], sum[1])) @@ -432,10 +424,8 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt, ff_ac3_quantize_mantissas(s); - if ((ret = ff_alloc_packet(avpkt, s->frame_size))) { - av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size, 0)) < 0) return ret; - } ff_ac3_output_frame(s, avpkt->data); if (frame->pts != AV_NOPTS_VALUE) |
