aacpsy.o aactab.o \
aacenc_is.o \
aacenc_tns.o \
+ aacenc_ltp.o \
aacenc_pred.o \
psymodel.o mpeg4audio.o kbdwin.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
typedef struct LongTermPrediction {
int8_t present;
int16_t lag;
+ int coef_idx;
INTFLOAT coef;
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
DECLARE_ALIGNED(32, INTFLOAT, saved)[1536]; ///< overlap
DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048]; ///< PCM output buffer
DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
+ DECLARE_ALIGNED(32, AAC_FLOAT, lcoeffs)[1024]; ///< MDCT of LTP coefficients (used by encoder)
DECLARE_ALIGNED(32, AAC_FLOAT, prcoeffs)[1024]; ///< Main prediction coefs (used by encoder)
PredictorState predictor_state[MAX_PREDICTORS];
INTFLOAT *ret; ///< PCM output
#include "aacenc_is.h"
#include "aacenc_tns.h"
+#include "aacenc_ltp.h"
#include "aacenc_pred.h"
#include "libavcodec/aaccoder_twoloop.h"
encode_window_bands_info,
quantize_and_encode_band,
ff_aac_encode_tns_info,
+ ff_aac_encode_ltp_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_pred,
+ ff_aac_adjust_common_ltp,
ff_aac_apply_main_pred,
ff_aac_apply_tns,
+ ff_aac_update_ltp,
+ ff_aac_ltp_insert_new_frame,
set_special_band_scalefactors,
search_for_pns,
mark_pns,
ff_aac_search_for_tns,
+ ff_aac_search_for_ltp,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
encode_window_bands_info,
quantize_and_encode_band,
ff_aac_encode_tns_info,
+ ff_aac_encode_ltp_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_pred,
+ ff_aac_adjust_common_ltp,
ff_aac_apply_main_pred,
ff_aac_apply_tns,
+ ff_aac_update_ltp,
+ ff_aac_ltp_insert_new_frame,
set_special_band_scalefactors,
search_for_pns,
mark_pns,
ff_aac_search_for_tns,
+ ff_aac_search_for_ltp,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
codebook_trellis_rate,
quantize_and_encode_band,
ff_aac_encode_tns_info,
+ ff_aac_encode_ltp_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_pred,
+ ff_aac_adjust_common_ltp,
ff_aac_apply_main_pred,
ff_aac_apply_tns,
+ ff_aac_update_ltp,
+ ff_aac_ltp_insert_new_frame,
set_special_band_scalefactors,
search_for_pns,
mark_pns,
ff_aac_search_for_tns,
+ ff_aac_search_for_ltp,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
encode_window_bands_info,
quantize_and_encode_band,
ff_aac_encode_tns_info,
+ ff_aac_encode_ltp_info,
ff_aac_encode_main_pred,
ff_aac_adjust_common_pred,
+ ff_aac_adjust_common_ltp,
ff_aac_apply_main_pred,
ff_aac_apply_tns,
+ ff_aac_update_ltp,
+ ff_aac_ltp_insert_new_frame,
set_special_band_scalefactors,
search_for_pns,
mark_pns,
ff_aac_search_for_tns,
+ ff_aac_search_for_ltp,
search_for_ms,
ff_aac_search_for_is,
ff_aac_search_for_pred,
.mid_side = 0,
.pns = 1,
.tns = 0,
+ .ltp = OPT_BANNED,
.pred = OPT_REQUIRED,
.intensity_stereo = 1,
},
.mid_side = 0,
.pns = 1,
.tns = 0,
+ .ltp = OPT_NEEDS_LTP,
.pred = OPT_NEEDS_MAIN,
.intensity_stereo = 1,
},
.mid_side = 0,
.pns = OPT_BANNED,
.tns = 0,
+ .ltp = OPT_BANNED,
+ .pred = OPT_BANNED,
+ .intensity_stereo = 1,
+ },
+ },
+ {FF_PROFILE_AAC_LTP,
+ { /* Long term prediction profile */
+ .mid_side = 0,
+ .pns = 1,
+ .tns = 0,
+ .ltp = OPT_REQUIRED,
.pred = OPT_BANNED,
.intensity_stereo = 1,
},
put_ics_info(s, &sce->ics);
if (s->coder->encode_main_pred)
s->coder->encode_main_pred(s, sce);
+ if (s->coder->encode_ltp_info)
+ s->coder->encode_ltp_info(s, sce, 0);
}
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
}
apply_window_and_mdct(s, sce, overlap);
+
+ if (s->options.ltp && s->coder->update_ltp) {
+ s->coder->update_ltp(s, sce);
+ apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
+ s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
+ }
+
if (isnan(cpe->ch->coeffs[0])) {
av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
return AVERROR(EINVAL);
sce = &cpe->ch[ch];
coeffs[ch] = sce->coeffs;
sce->ics.predictor_present = 0;
+ sce->ics.ltp.present = 0;
+ memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
for (w = 0; w < 128; w++)
apply_mid_side_stereo(cpe);
}
adjust_frame_information(cpe, chans);
+ if (s->options.ltp) { /* LTP */
+ for (ch = 0; ch < chans; ch++) {
+ sce = &cpe->ch[ch];
+ s->cur_channel = start_ch + ch;
+ if (s->coder->search_for_ltp)
+ s->coder->search_for_ltp(s, sce, cpe->common_window);
+ if (sce->ics.ltp.present) pred_mode = 1;
+ }
+ s->cur_channel = start_ch;
+ if (s->coder->adjust_common_ltp)
+ s->coder->adjust_common_ltp(s, cpe);
+ }
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]);
+ if (s->coder->encode_ltp_info)
+ s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
encode_ms_info(&s->pb, cpe);
if (cpe->ms_mode) ms_mode = 1;
}
}
} while (1);
+ if (s->options.ltp && s->coder->ltp_insert_new_frame)
+ s->coder->ltp_insert_new_frame(s);
+
put_bits(&s->pb, 3, TYPE_END);
flush_put_bits(&s->pb);
avctx->frame_bits = put_bits_count(&s->pb);
AAC_OPT_SET(&s->options, p_opt, 1, coder);
AAC_OPT_SET(&s->options, p_opt, 0, pns);
AAC_OPT_SET(&s->options, p_opt, 1, tns);
+ AAC_OPT_SET(&s->options, p_opt, 0, ltp);
AAC_OPT_SET(&s->options, p_opt, 0, pred);
AAC_OPT_SET(&s->options, p_opt, 1, mid_side);
AAC_OPT_SET(&s->options, p_opt, 0, intensity_stereo);
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
{"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
{"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
+ {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
{"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
{NULL}
};
int coder;
int pns;
int tns;
+ int ltp;
int pred;
int mid_side;
int intensity_stereo;
void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
int scale_idx, int cb, const float lambda, int rtz);
void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
+ void (*encode_ltp_info)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
void (*adjust_common_pred)(struct AACEncContext *s, ChannelElement *cpe);
+ void (*adjust_common_ltp)(struct AACEncContext *s, ChannelElement *cpe);
void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
+ void (*update_ltp)(struct AACEncContext *s, SingleChannelElement *sce);
+ void (*ltp_insert_new_frame)(struct AACEncContext *s);
void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
void (*mark_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
+ void (*search_for_ltp)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
--- /dev/null
+/*
+ * AAC encoder TNS
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder long term prediction
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include "aacenc_ltp.h"
+#include "aacenc_quantization.h"
+#include "aacenc_utils.h"
+
+/**
+ * Encode LTP data.
+ */
+void ff_aac_encode_ltp_info(AACEncContext *s, SingleChannelElement *sce,
+ int common_window)
+{
+ int i;
+ IndividualChannelStream *ics = &sce->ics;
+ if (s->profile != FF_PROFILE_AAC_LTP || !ics->predictor_present)
+ return;
+ if (common_window)
+ put_bits(&s->pb, 1, 0);
+ put_bits(&s->pb, 1, ics->ltp.present);
+ if (!ics->ltp.present)
+ return;
+ put_bits(&s->pb, 11, ics->ltp.lag);
+ put_bits(&s->pb, 3, ics->ltp.coef_idx);
+ for (i = 0; i < FFMIN(ics->max_sfb, MAX_LTP_LONG_SFB); i++)
+ put_bits(&s->pb, 1, ics->ltp.used[i]);
+}
+
+void ff_aac_ltp_insert_new_frame(AACEncContext *s)
+{
+ int i, ch, tag, chans, cur_channel, start_ch = 0;
+ ChannelElement *cpe;
+ SingleChannelElement *sce;
+ for (i = 0; i < s->chan_map[0]; i++) {
+ cpe = &s->cpe[i];
+ tag = s->chan_map[i+1];
+ chans = tag == TYPE_CPE ? 2 : 1;
+ for (ch = 0; ch < chans; ch++) {
+ sce = &cpe->ch[ch];
+ cur_channel = start_ch + ch;
+ /* New sample + overlap */
+ memcpy(&sce->ltp_state[0], &sce->ltp_state[1024], 1024*sizeof(sce->ltp_state[0]));
+ memcpy(&sce->ltp_state[1024], &s->planar_samples[cur_channel][2048], 1024*sizeof(sce->ltp_state[0]));
+ memcpy(&sce->ltp_state[2048], &sce->ret_buf[0], 1024*sizeof(sce->ltp_state[0]));
+ }
+ start_ch += chans;
+ }
+}
+
+/**
+ * Process LTP parameters
+ * @see Patent WO2006070265A1
+ */
+void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce)
+{
+ int i, j, lag;
+ float corr, s0, s1, max_corr = 0.0f;
+ float *samples = &s->planar_samples[s->cur_channel][1024];
+ float *pred_signal = &sce->ltp_state[0];
+ int samples_num = 2048;
+
+ if (s->profile != FF_PROFILE_AAC_LTP)
+ return;
+
+ /* Calculate lag */
+ for (i = 0; i < samples_num; i++) {
+ s0 = s1 = 0.0f;
+ for (j = 0; j < samples_num; j++) {
+ if (j + 1024 < i)
+ continue;
+ s0 += samples[j]*pred_signal[j-i+1024];
+ s1 += pred_signal[j-i+1024]*pred_signal[j-i+1024];
+ }
+ corr = s1 > 0.0f ? s0/sqrt(s1) : 0.0f;
+ if (corr > max_corr) {
+ max_corr = corr;
+ lag = i;
+ }
+ }
+ lag = av_clip(lag, 0, 2048); /* 11 bits => 2^11 = 2048 */
+
+ if (!lag) {
+ sce->ics.ltp.lag = lag;
+ return;
+ }
+
+ s0 = s1 = 0.0f;
+ for (i = 0; i < lag; i++) {
+ s0 += samples[i];
+ s1 += pred_signal[i-lag+1024];
+ }
+
+ sce->ics.ltp.coef_idx = quant_array_idx(s0/s1, ltp_coef, 8);
+ sce->ics.ltp.coef = ltp_coef[sce->ics.ltp.coef_idx];
+
+ /* Predict the new samples */
+ if (lag < 1024)
+ samples_num = lag + 1024;
+ for (i = 0; i < samples_num; i++)
+ pred_signal[i+1024] = sce->ics.ltp.coef*pred_signal[i-lag+1024];
+ memset(&pred_signal[samples_num], 0, (2048 - samples_num)*sizeof(float));
+
+ sce->ics.ltp.lag = lag;
+}
+
+void ff_aac_adjust_common_ltp(AACEncContext *s, ChannelElement *cpe)
+{
+ int sfb, count = 0;
+ SingleChannelElement *sce0 = &cpe->ch[0];
+ SingleChannelElement *sce1 = &cpe->ch[1];
+
+ if (!cpe->common_window ||
+ sce0->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
+ sce1->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE)
+ return;
+
+ for (sfb = 0; sfb < FFMIN(sce0->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++) {
+ int sum = sce0->ics.ltp.used[sfb] + sce1->ics.ltp.used[sfb];
+ if (sum != 2) {
+ sce0->ics.ltp.used[sfb] = 0;
+ } else if (sum == 2) {
+ count++;
+ }
+ }
+
+ sce0->ics.ltp.present = !!count;
+ sce0->ics.predictor_present = !!count;
+}
+
+/**
+ * Mark LTP sfb's
+ */
+void ff_aac_search_for_ltp(AACEncContext *s, SingleChannelElement *sce,
+ int common_window)
+{
+ int w, g, w2, i, start = 0, count = 0;
+ int saved_bits = -(15 + FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB));
+ float *C34 = &s->scoefs[128*0], *PCD = &s->scoefs[128*1];
+ float *PCD34 = &s->scoefs[128*2];
+ const int max_ltp = FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB);
+
+ if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
+ !sce->ics.ltp.lag)
+ return;
+
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+ start = 0;
+ for (g = 0; g < sce->ics.num_swb; g++) {
+ int bits1 = 0, bits2 = 0;
+ float dist1 = 0.0f, dist2 = 0.0f;
+ if (w*16+g > max_ltp) {
+ start += sce->ics.swb_sizes[g];
+ continue;
+ }
+ for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+ int bits_tmp1, bits_tmp2;
+ FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+ for (i = 0; i < sce->ics.swb_sizes[g]; i++)
+ PCD[i] = sce->coeffs[start+(w+w2)*128+i] - sce->lcoeffs[start+(w+w2)*128+i];
+ abs_pow34_v(C34, &sce->coeffs[start+(w+w2)*128], sce->ics.swb_sizes[g]);
+ abs_pow34_v(PCD34, PCD, sce->ics.swb_sizes[g]);
+ dist1 += quantize_band_cost(s, &sce->coeffs[start+(w+w2)*128], C34, sce->ics.swb_sizes[g],
+ sce->sf_idx[(w+w2)*16+g], sce->band_type[(w+w2)*16+g],
+ s->lambda/band->threshold, INFINITY, &bits_tmp1, NULL, 0);
+ dist2 += quantize_band_cost(s, PCD, PCD34, sce->ics.swb_sizes[g],
+ sce->sf_idx[(w+w2)*16+g],
+ sce->band_type[(w+w2)*16+g],
+ s->lambda/band->threshold, INFINITY, &bits_tmp2, NULL, 0);
+ bits1 += bits_tmp1;
+ bits2 += bits_tmp2;
+ }
+ if (dist2 < dist1 && bits2 < bits1) {
+ for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
+ for (i = 0; i < sce->ics.swb_sizes[g]; i++)
+ sce->coeffs[start+(w+w2)*128+i] -= sce->lcoeffs[start+(w+w2)*128+i];
+ sce->ics.ltp.used[w*16+g] = 1;
+ saved_bits += bits1 - bits2;
+ count++;
+ }
+ start += sce->ics.swb_sizes[g];
+ }
+ }
+
+ sce->ics.ltp.present = !!count && (saved_bits >= 0);
+ sce->ics.predictor_present = !!sce->ics.ltp.present;
+
+ /* Reset any marked sfbs */
+ if (!sce->ics.ltp.present && !!count) {
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+ start = 0;
+ for (g = 0; g < sce->ics.num_swb; g++) {
+ if (sce->ics.ltp.used[w*16+g]) {
+ for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+ for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
+ sce->coeffs[start+(w+w2)*128+i] += sce->lcoeffs[start+(w+w2)*128+i];
+ }
+ }
+ }
+ start += sce->ics.swb_sizes[g];
+ }
+ }
+ }
+}
--- /dev/null
+/*
+ * AAC encoder long term prediction extension
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder long term prediction extension
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_LTP_H
+#define AVCODEC_AACENC_LTP_H
+
+#include "aacenc.h"
+
+void ff_aac_encode_ltp_info(AACEncContext *s, SingleChannelElement *sce,
+ int common_window);
+void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_adjust_common_ltp(AACEncContext *s, ChannelElement *cpe);
+void ff_aac_ltp_insert_new_frame(AACEncContext *s);
+void ff_aac_search_for_ltp(AACEncContext *s, SingleChannelElement *sce,
+ int common_window);
+
+#endif /* AVCODEC_AACENC_LTP_H */
projects / ffmpeg / commitdiff