*/
/**
- * @file libavcodec/aacenc.c
+ * @file
* AAC encoder
*/
#include "psymodel.h"
+#define AAC_MAX_CHANNELS 6
+
static const uint8_t swb_size_1024_96[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
return -1;
}
- if (avctx->channels > 6) {
+ if (avctx->channels > AAC_MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
return -1;
}
av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
return -1;
}
+ if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
+ av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
+ return -1;
+ }
s->samplerate_index = i;
dsputil_init(&s->dsp, avctx);
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]);
- avctx->extradata = av_malloc(2);
+ avctx->extradata = av_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
lengths[1] = ff_aac_num_swb_128[i];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
s->psypp = ff_psy_preprocess_init(avctx);
- s->coder = &ff_aac_coders[0];
+ s->coder = &ff_aac_coders[2];
s->lambda = avctx->global_quality ? avctx->global_quality : 120;
-#if !CONFIG_HARDCODED_TABLES
- for (i = 0; i < 428; i++)
- ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
-#endif /* CONFIG_HARDCODED_TABLES */
- if (avctx->channels > 5)
- av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
- "The output will most likely be an illegal bitstream.\n");
+ ff_aac_tableinit();
return 0;
}
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
- SingleChannelElement *sce, short *audio, int channel)
+ SingleChannelElement *sce, short *audio)
{
- int i, j, k;
+ int i, k;
+ const int chans = avctx->channels;
const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
s->output[i] = sce->saved[i];
}
if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
- j = channel;
- for (i = 0; i < 1024; i++, j += avctx->channels) {
- s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
- sce->saved[i] = audio[j] * lwindow[i];
+ for (i = 0; i < 1024; i++) {
+ s->output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
+ sce->saved[i] = audio[i * chans] * lwindow[i];
}
} else {
- j = channel;
- for (i = 0; i < 448; i++, j += avctx->channels)
- s->output[i+1024] = audio[j];
- for (i = 448; i < 576; i++, j += avctx->channels)
- s->output[i+1024] = audio[j] * swindow[576 - i - 1];
+ for (i = 0; i < 448; i++)
+ s->output[i+1024] = audio[i * chans];
+ for (; i < 576; i++)
+ s->output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
- j = channel;
- for (i = 0; i < 1024; i++, j += avctx->channels)
- sce->saved[i] = audio[j];
+ for (i = 0; i < 1024; i++)
+ sce->saved[i] = audio[i * chans];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
} else {
- j = channel;
for (k = 0; k < 1024; k += 128) {
for (i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
? sce->saved[i]
- : audio[channel + (i-1024)*avctx->channels];
+ : audio[(i-1024)*chans];
s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
}
- j = channel;
- for (i = 0; i < 1024; i++, j += avctx->channels)
- sce->saved[i] = audio[j];
+ for (i = 0; i < 1024; i++)
+ sce->saved[i] = audio[i * chans];
}
}
static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
{
int i, w, w2, g, ch;
- int start, sum, maxsfb, cmaxsfb;
+ int start, maxsfb, cmaxsfb;
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
cpe->ch[ch].pulse.num_pulse = 0;
for (w = 0; w < ics->num_windows*16; w += 16) {
for (g = 0; g < ics->num_swb; g++) {
- sum = 0;
//apply M/S
- if (!ch && cpe->ms_mask[w + g]) {
+ if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
for (i = 0; i < ics->swb_sizes[g]; i++) {
cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
int i, j, chans, tag, start_ch;
const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
int chan_el_counter[4];
- FFPsyWindowInfo windows[avctx->channels];
+ FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
if (s->last_frame)
return 0;
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
- samples2 = samples + start_ch;
- la = samples2 + 1024 * avctx->channels + start_ch;
- if (!data)
- la = NULL;
for (j = 0; j < chans; j++) {
IndividualChannelStream *ics = &cpe->ch[j].ics;
int k;
- wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
+ int cur_channel = start_ch + j;
+ samples2 = samples + cur_channel;
+ la = samples2 + (448+64) * avctx->channels;
+ if (!data)
+ la = NULL;
+ if (tag == TYPE_LFE) {
+ wi[j].window_type[0] = ONLY_LONG_SEQUENCE;
+ wi[j].window_shape = 0;
+ wi[j].num_windows = 1;
+ wi[j].grouping[0] = 1;
+ } else {
+ wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, cur_channel,
+ ics->window_sequence[0]);
+ }
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = wi[j].window_type[0];
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = wi[j].window_shape;
ics->num_windows = wi[j].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
- ics->num_swb = s->psy.num_bands[ics->num_windows == 8];
+ ics->num_swb = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8];
for (k = 0; k < ics->num_windows; k++)
ics->group_len[k] = wi[j].grouping[k];
- s->cur_channel = start_ch + j;
- apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
+ apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2);
}
start_ch += chans;
}
tag = 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 (j = 0; j < chans; j++) {
+ s->cur_channel = start_ch + j;
+ ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
}
cpe->common_window = 0;
}
}
}
+ s->cur_channel = start_ch;
if (cpe->common_window && s->coder->search_for_ms)
s->coder->search_for_ms(s, cpe, s->lambda);
adjust_frame_information(s, cpe, chans);
- put_bits(&s->pb, 3, tag);
- put_bits(&s->pb, 4, chan_el_counter[tag]++);
if (chans == 2) {
put_bits(&s->pb, 1, cpe->common_window);
if (cpe->common_window) {
}
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
- ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;
aac_encode_init,
aac_encode_frame,
aac_encode_end,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};