/***********************************
* TODOs:
* add sane pulse detection
- * add temporal noise shaping
***********************************/
#include "libavutil/float_dsp.h"
#include "aac.h"
#include "aactab.h"
#include "aacenc.h"
+#include "aacenctab.h"
+#include "aacenc_utils.h"
#include "psymodel.h"
-#define AAC_MAX_CHANNELS 6
-
-#define ERROR_IF(cond, ...) \
- if (cond) { \
- av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
- return AVERROR(EINVAL); \
- }
-
-#define WARN_IF(cond, ...) \
- if (cond) { \
- av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
- }
-
-
-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,
- 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_64[] = {
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
- 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
- 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
-};
-
-static const uint8_t swb_size_1024_48[] = {
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
- 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
- 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
- 96
-};
-
-static const uint8_t swb_size_1024_32[] = {
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
- 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
- 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
-};
-
-static const uint8_t swb_size_1024_24[] = {
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
- 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
- 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_16[] = {
- 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
- 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
- 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_8[] = {
- 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
- 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
- 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
-};
-
-static const uint8_t *swb_size_1024[] = {
- swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
- swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
- swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
- swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
- swb_size_1024_8
-};
-
-static const uint8_t swb_size_128_96[] = {
- 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
-};
-
-static const uint8_t swb_size_128_48[] = {
- 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
-};
-
-static const uint8_t swb_size_128_24[] = {
- 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
-};
-
-static const uint8_t swb_size_128_16[] = {
- 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
-};
-
-static const uint8_t swb_size_128_8[] = {
- 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
-};
-
-static const uint8_t *swb_size_128[] = {
- /* the last entry on the following row is swb_size_128_64 but is a
- duplicate of swb_size_128_96 */
- swb_size_128_96, swb_size_128_96, swb_size_128_96,
- swb_size_128_48, swb_size_128_48, swb_size_128_48,
- swb_size_128_24, swb_size_128_24, swb_size_128_16,
- swb_size_128_16, swb_size_128_16, swb_size_128_8,
- swb_size_128_8
-};
-
-/** default channel configurations */
-static const uint8_t aac_chan_configs[6][5] = {
- {1, TYPE_SCE}, // 1 channel - single channel element
- {1, TYPE_CPE}, // 2 channels - channel pair
- {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
- {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
- {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
- {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
-};
-
-/**
- * Table to remap channels from libavcodec's default order to AAC order.
- */
-static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][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;
init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
- put_bits(&pb, 5, 2); //object type - AAC-LC
+ put_bits(&pb, 5, s->profile+1); //profile
put_bits(&pb, 4, s->samplerate_index); //sample rate index
put_bits(&pb, 4, s->channels);
//GASpecificConfig
s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
else
for (i = 0; i < 1024; i += 128)
- s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
+ s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
}
put_bits(&s->pb, 1, info->use_kb_window[0]);
if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
put_bits(&s->pb, 6, info->max_sfb);
- put_bits(&s->pb, 1, 0); // no prediction
+ put_bits(&s->pb, 1, !!info->predictor_present);
} else {
put_bits(&s->pb, 4, info->max_sfb);
for (w = 1; w < 8; w++)
{
int i, w, w2, g, ch;
int maxsfb, cmaxsfb;
- IndividualChannelStream *ics;
-
- if (cpe->common_window) {
- ics = &cpe->ch[0].ics;
- for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
- for (w2 = 0; w2 < ics->group_len[w]; w2++) {
- int start = (w+w2) * 128;
- for (g = 0; g < ics->num_swb; g++) {
- //apply Intensity stereo coeffs transformation
- if (cpe->is_mask[w*16 + g]) {
- int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
- float scale = cpe->ch[0].is_ener[w*16+g];
- for (i = 0; i < ics->swb_sizes[g]; i++) {
- cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + p*cpe->ch[1].pcoeffs[start+i]) * scale;
- cpe->ch[1].coeffs[start+i] = 0.0f;
- }
- } else if (cpe->ms_mask[w*16 + g] &&
- cpe->ch[0].band_type[w*16 + g] < NOISE_BT &&
- cpe->ch[1].band_type[w*16 + g] < NOISE_BT) {
- for (i = 0; i < ics->swb_sizes[g]; i++) {
- cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + cpe->ch[1].pcoeffs[start+i]) * 0.5f;
- cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].pcoeffs[start+i];
- }
- }
- start += ics->swb_sizes[g];
- }
- }
- }
- }
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
}
}
+static void apply_intensity_stereo(ChannelElement *cpe)
+{
+ int w, w2, g, i;
+ IndividualChannelStream *ics = &cpe->ch[0].ics;
+ if (!cpe->common_window)
+ return;
+ for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
+ for (w2 = 0; w2 < ics->group_len[w]; w2++) {
+ int start = (w+w2) * 128;
+ for (g = 0; g < ics->num_swb; g++) {
+ int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
+ float scale = cpe->ch[0].is_ener[w*16+g];
+ if (!cpe->is_mask[w*16 + g]) {
+ start += ics->swb_sizes[g];
+ continue;
+ }
+ for (i = 0; i < ics->swb_sizes[g]; i++) {
+ float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
+ cpe->ch[0].coeffs[start+i] = sum;
+ cpe->ch[1].coeffs[start+i] = 0.0f;
+ }
+ start += ics->swb_sizes[g];
+ }
+ }
+ }
+}
+
+static void apply_mid_side_stereo(ChannelElement *cpe)
+{
+ int w, w2, g, i;
+ IndividualChannelStream *ics = &cpe->ch[0].ics;
+ if (!cpe->common_window)
+ return;
+ for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
+ for (w2 = 0; w2 < ics->group_len[w]; w2++) {
+ int start = (w+w2) * 128;
+ for (g = 0; g < ics->num_swb; g++) {
+ if (!cpe->ms_mask[w*16 + g]) {
+ start += ics->swb_sizes[g];
+ continue;
+ }
+ for (i = 0; i < ics->swb_sizes[g]; i++) {
+ float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
+ float R = L - cpe->ch[1].coeffs[start+i];
+ cpe->ch[0].coeffs[start+i] = L;
+ cpe->ch[1].coeffs[start+i] = R;
+ }
+ start += ics->swb_sizes[g];
+ }
+ }
+ }
+}
+
/**
* Encode scalefactor band coding type.
*/
{
int w;
+ if (s->coder->set_special_band_scalefactors)
+ s->coder->set_special_band_scalefactors(s, sce);
+
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
}
start += sce->ics.swb_sizes[i];
continue;
}
- for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
- s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
- sce->ics.swb_sizes[i],
+ for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
+ s->coder->quantize_and_encode_band(s, &s->pb,
+ &sce->coeffs[start + w2*128],
+ NULL, sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
- s->lambda, sce->ics.window_clipping[w]);
+ s->lambda,
+ sce->ics.window_clipping[w]);
+ }
start += sce->ics.swb_sizes[i];
}
}
for (w = 0; w < sce->ics.num_windows; w++) {
start = 0;
for (i = 0; i < sce->ics.max_sfb; i++) {
- float *swb_coeffs = sce->coeffs + start + w*128;
+ float *swb_coeffs = &sce->coeffs[start + w*128];
for (j = 0; j < sce->ics.swb_sizes[i]; j++)
swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
start += sce->ics.swb_sizes[i];
int common_window)
{
put_bits(&s->pb, 8, sce->sf_idx[0]);
- if (!common_window)
+ if (!common_window) {
put_ics_info(s, &sce->ics);
+ if (s->coder->encode_main_pred)
+ s->coder->encode_main_pred(s, sce);
+ }
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
encode_pulses(s, &sce->pulse);
- put_bits(&s->pb, 1, 0); //tns
+ put_bits(&s->pb, 1, !!sce->tns.present);
+ if (s->coder->encode_tns_info)
+ s->coder->encode_tns_info(s, sce);
put_bits(&s->pb, 1, 0); //ssr
encode_spectral_coeffs(s, sce);
return 0;
AACEncContext *s = avctx->priv_data;
float **samples = s->planar_samples, *samples2, *la, *overlap;
ChannelElement *cpe;
- int i, ch, w, g, chans, tag, start_ch, ret, ms_mode = 0, is_mode = 0;
+ SingleChannelElement *sce;
+ int i, ch, w, chans, tag, start_ch, ret;
+ int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
int chan_el_counter[4];
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
ics->num_windows = wi[ch].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
+ ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
+ ff_swb_offset_128 [s->samplerate_index]:
+ ff_swb_offset_1024[s->samplerate_index];
+ ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
+ ff_tns_max_bands_128 [s->samplerate_index]:
+ ff_tns_max_bands_1024[s->samplerate_index];
clip_avoidance_factor = 0.0f;
for (w = 0; w < ics->num_windows; w++)
ics->group_len[w] = wi[ch].grouping[w];
tag = s->chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
+ cpe->common_window = 0;
memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
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;
+ for (ch = 0; ch < chans; ch++) {
+ sce = &cpe->ch[ch];
+ coeffs[ch] = sce->coeffs;
+ sce->ics.predictor_present = 0;
+ memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
+ memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
+ for (w = 0; w < 128; w++)
+ if (sce->band_type[w] > RESERVED_BT)
+ sce->band_type[w] = 0;
+ }
s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
for (ch = 0; ch < chans; ch++) {
s->cur_channel = start_ch + ch;
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
}
- cpe->common_window = 0;
if (chans > 1
&& wi[0].window_type[0] == wi[1].window_type[0]
&& wi[0].window_shape == wi[1].window_shape) {
}
}
}
- if (s->options.pns && s->coder->search_for_pns) {
+ for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
+ sce = &cpe->ch[ch];
+ s->cur_channel = start_ch + ch;
+ if (s->options.pns && s->coder->search_for_pns)
+ s->coder->search_for_pns(s, avctx, sce);
+ if (s->options.tns && s->coder->search_for_tns)
+ s->coder->search_for_tns(s, sce);
+ if (s->options.tns && s->coder->apply_tns_filt)
+ s->coder->apply_tns_filt(s, sce);
+ if (sce->tns.present)
+ tns_mode = 1;
+ }
+ s->cur_channel = start_ch;
+ if (s->options.intensity_stereo) { /* Intensity Stereo */
+ if (s->coder->search_for_is)
+ s->coder->search_for_is(s, avctx, cpe);
+ if (cpe->is_mode) is_mode = 1;
+ apply_intensity_stereo(cpe);
+ }
+ if (s->options.pred) { /* Prediction */
for (ch = 0; ch < chans; ch++) {
+ sce = &cpe->ch[ch];
s->cur_channel = start_ch + ch;
- s->coder->search_for_pns(s, avctx, &cpe->ch[ch]);
+ if (s->options.pred && s->coder->search_for_pred)
+ s->coder->search_for_pred(s, sce);
+ if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
}
- }
- s->cur_channel = start_ch;
- if (s->options.stereo_mode && cpe->common_window) {
- if (s->options.stereo_mode > 0) {
- IndividualChannelStream *ics = &cpe->ch[0].ics;
- for (w = 0; w < ics->num_windows; w += ics->group_len[w])
- for (g = 0; g < ics->num_swb; g++)
- cpe->ms_mask[w*16+g] = 1;
- } else if (s->coder->search_for_ms) {
- s->coder->search_for_ms(s, cpe);
+ if (s->coder->adjust_common_prediction)
+ s->coder->adjust_common_prediction(s, cpe);
+ for (ch = 0; ch < chans; ch++) {
+ sce = &cpe->ch[ch];
+ s->cur_channel = start_ch + ch;
+ if (s->options.pred && s->coder->apply_main_pred)
+ s->coder->apply_main_pred(s, sce);
}
+ s->cur_channel = start_ch;
}
- if (chans > 1 && s->options.intensity_stereo && s->coder->search_for_is) {
- s->coder->search_for_is(s, avctx, cpe);
- if (cpe->is_mode) is_mode = 1;
+ if (s->options.stereo_mode) { /* Mid/Side stereo */
+ if (s->options.stereo_mode == -1 && s->coder->search_for_ms)
+ s->coder->search_for_ms(s, cpe);
+ else if (cpe->common_window)
+ memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
+ for (w = 0; w < 128; w++)
+ cpe->ms_mask[w] = cpe->is_mask[w] ? 0 : cpe->ms_mask[w];
+ apply_mid_side_stereo(cpe);
}
- if (s->coder->set_special_band_scalefactors)
- for (ch = 0; ch < chans; ch++)
- s->coder->set_special_band_scalefactors(s, &cpe->ch[ch]);
adjust_frame_information(cpe, chans);
if (chans == 2) {
put_bits(&s->pb, 1, cpe->common_window);
if (cpe->common_window) {
put_ics_info(s, &cpe->ch[0].ics);
+ if (s->coder->encode_main_pred)
+ s->coder->encode_main_pred(s, &cpe->ch[0]);
encode_ms_info(&s->pb, cpe);
if (cpe->ms_mode) ms_mode = 1;
}
s->psy.bitres.bits = frame_bits / s->channels;
break;
}
- if (is_mode || ms_mode) {
+ if (is_mode || ms_mode || tns_mode || pred_mode) {
for (i = 0; i < s->chan_map[0]; i++) {
// Must restore coeffs
chans = tag == TYPE_CPE ? 2 : 1;
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
ff_psy_end(&s->psy);
+ ff_lpc_end(&s->lpc);
if (s->psypp)
ff_psy_preprocess_end(s->psypp);
av_freep(&s->buffer.samples);
s->channels = avctx->channels;
- ERROR_IF(i == 16
- || i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024))
- || i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)),
+ ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
"Unsupported sample rate %d\n", avctx->sample_rate);
ERROR_IF(s->channels > AAC_MAX_CHANNELS,
"Unsupported number of channels: %d\n", s->channels);
- ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
- "Unsupported profile %d\n", avctx->profile);
WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
"Too many bits per frame requested, clamping to max\n");
+ if (avctx->profile == FF_PROFILE_AAC_MAIN) {
+ s->options.pred = 1;
+ } else if ((avctx->profile == FF_PROFILE_AAC_LOW ||
+ avctx->profile == FF_PROFILE_UNKNOWN) && s->options.pred) {
+ s->profile = 0; /* Main */
+ WARN_IF(1, "Prediction requested, changing profile to AAC-Main\n");
+ } else if (avctx->profile == FF_PROFILE_AAC_LOW ||
+ avctx->profile == FF_PROFILE_UNKNOWN) {
+ s->profile = 1; /* Low */
+ } else {
+ ERROR_IF(1, "Unsupported profile %d\n", avctx->profile);
+ }
+
+ if (s->options.aac_coder != AAC_CODER_TWOLOOP) {
+ s->options.intensity_stereo = 0;
+ s->options.pns = 0;
+ }
avctx->bit_rate = (int)FFMIN(
6144 * s->channels / 1024.0 * avctx->sample_rate,
avctx->extradata_size = 5;
put_audio_specific_config(avctx);
- sizes[0] = swb_size_1024[i];
- sizes[1] = swb_size_128[i];
+ sizes[0] = ff_aac_swb_size_1024[i];
+ sizes[1] = ff_aac_swb_size_128[i];
lengths[0] = ff_aac_num_swb_1024[i];
lengths[1] = ff_aac_num_swb_128[i];
for (i = 0; i < s->chan_map[0]; i++)
goto fail;
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[s->options.aac_coder];
+ ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
if (HAVE_MIPSDSPR1)
ff_aac_coder_init_mips(s);
{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
{"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
- {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},
+ {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "aac_pns"},
{"disable", "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
{"enable", "Enable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
- {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
+ {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
{"disable", "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
{"enable", "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
+ {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_tns"},
+ {"disable", "Disable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
+ {"enable", "Enable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
+ {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pred"},
+ {"disable", "Disable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
+ {"enable", "Enable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
{NULL}
};
LIBAVUTIL_VERSION_INT,
};
-/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
- * failures */
-static const int mpeg4audio_sample_rates[16] = {
- 96000, 88200, 64000, 48000, 44100, 32000,
- 24000, 22050, 16000, 12000, 11025, 8000, 7350
-};
-
AVCodec ff_aac_encoder = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),