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;
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows(7);
+ s->chan_map = aac_chan_configs[avctx->channels-1];
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
- s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
+ s->cpe = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]);
avctx->extradata = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE);
avctx->extradata_size = 5;
put_audio_specific_config(avctx);
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);
+ 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];
if (msc == 0 || ics0->max_sfb == 0)
cpe->ms_mode = 0;
else
- cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
+ cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
}
}
int16_t *samples = s->samples, *samples2, *la;
ChannelElement *cpe;
int i, ch, w, g, chans, tag, start_ch;
- const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
int chan_el_counter[4];
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
} else {
start_ch = 0;
samples2 = s->samples + 1024 * avctx->channels;
- for (i = 0; i < chan_map[0]; i++) {
- tag = chan_map[i+1];
+ 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 + channel_maps[avctx->channels-1][start_ch],
}
start_ch = 0;
- for (i = 0; i < chan_map[0]; i++) {
+ for (i = 0; i < s->chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
- tag = chan_map[i+1];
+ tag = s->chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
for (ch = 0; ch < chans; ch++) {
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];
put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
start_ch = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
- for (i = 0; i < chan_map[0]; i++) {
+ for (i = 0; i < s->chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
- tag = chan_map[i+1];
+ const float *coeffs[2];
+ tag = s->chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
put_bits(&s->pb, 3, tag);
put_bits(&s->pb, 4, chan_el_counter[tag]++);
+ for (ch = 0; ch < chans; ch++)
+ coeffs[ch] = cpe->ch[ch].coeffs;
+ s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
for (ch = 0; ch < chans; ch++) {
- s->cur_channel = start_ch + ch;
- s->psy.model->analyze(&s->psy, s->cur_channel, cpe->ch[ch].coeffs, &wi[ch]);
+ s->cur_channel = start_ch * 2 + ch;
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
}
cpe->common_window = 0;
}
}
}
- s->cur_channel = start_ch;
+ s->cur_channel = start_ch * 2;
if (s->options.stereo_mode && cpe->common_window) {
if (s->options.stereo_mode > 0) {
IndividualChannelStream *ics = &cpe->ch[0].ics;
};
AVCodec ff_aac_encoder = {
- "aac",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_AAC,
- sizeof(AACEncContext),
- aac_encode_init,
- aac_encode_frame,
- aac_encode_end,
+ .name = "aac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_AAC,
+ .priv_data_size = sizeof(AACEncContext),
+ .init = aac_encode_init,
+ .encode = aac_encode_frame,
+ .close = aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),