* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
- * 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
*/
{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
};
+static const uint8_t channel_maps[][AAC_MAX_CHANNELS] = {
+ { 0 },
+ { 0, 1 },
+ { 2, 0, 1 },
+ { 2, 0, 1, 3 },
+ { 2, 0, 1, 3, 4 },
+ { 2, 0, 1, 4, 5, 3 },
+};
+
/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
AACEncContext *s = avctx->priv_data;
int i;
const uint8_t *sizes[2];
+ uint8_t grouping[AAC_MAX_CHANNELS];
int lengths[2];
avctx->frame_size = 1024;
sizes[1] = swb_size_128[i];
lengths[0] = ff_aac_num_swb_1024[i];
lengths[1] = ff_aac_num_swb_128[i];
- ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], &s->chan_map[1]);
+ for (i = 0; i < s->chan_map[0]; i++)
+ grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
+ ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[2];
return 0;
if (data) {
if (!s->psypp) {
- memcpy(s->samples + 1024 * avctx->channels, data,
- 1024 * avctx->channels * sizeof(s->samples[0]));
+ if (avctx->channels <= 2) {
+ memcpy(s->samples + 1024 * avctx->channels, data,
+ 1024 * avctx->channels * sizeof(s->samples[0]));
+ } else {
+ for (i = 0; i < 1024; i++)
+ for (ch = 0; ch < avctx->channels; ch++)
+ s->samples[(i + 1024) * avctx->channels + ch] =
+ ((int16_t*)data)[i * avctx->channels +
+ channel_maps[avctx->channels-1][ch]];
+ }
} else {
start_ch = 0;
samples2 = s->samples + 1024 * avctx->channels;
for (i = 0; i < s->chan_map[0]; i++) {
tag = s->chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
- ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
+ ff_psy_preprocess(s->psypp,
+ (uint16_t*)data + channel_maps[avctx->channels-1][start_ch],
samples2 + start_ch, start_ch, chans);
start_ch += chans;
}
wi[ch].window_shape = 0;
wi[ch].num_windows = 1;
wi[ch].grouping[0] = 1;
+
+ /* Only the lowest 12 coefficients are used in a LFE channel.
+ * The expression below results in only the bottom 8 coefficients
+ * being used for 11.025kHz to 16kHz sample rates.
+ */
+ ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
} else {
wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
ics->window_sequence[0]);
ics->use_kb_window[0] = wi[ch].window_shape;
ics->num_windows = wi[ch].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
- ics->num_swb = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8];
+ ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
for (w = 0; w < ics->num_windows; w++)
ics->group_len[w] = wi[ch].grouping[w];