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-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/aac.h1
-rw-r--r--libavcodec/aaccoder.c9
-rw-r--r--libavcodec/aacenc.c23
-rw-r--r--libavcodec/aacenc.h3
-rw-r--r--libavcodec/aacenc_tns.c235
-rw-r--r--libavcodec/aacenc_tns.h45
7 files changed, 311 insertions, 6 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 3d6913a368..a1f99e4616 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -133,6 +133,7 @@ OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacenctab.o \
aacpsy.o aactab.o \
aacenc_is.o \
+ aacenc_tns.o \
psymodel.o mpeg4audio.o kbdwin.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index 4ac4a34a48..9ab2639ac8 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -195,6 +195,7 @@ typedef struct TemporalNoiseShaping {
int length[8][4];
int direction[8][4];
int order[8][4];
+ int coef_idx[8][4][TNS_MAX_ORDER];
INTFLOAT coef[8][4][TNS_MAX_ORDER];
} TemporalNoiseShaping;
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c
index 5df30a61ef..911668a42a 100644
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -45,6 +45,7 @@
#include "aac_tablegen_decl.h"
#include "aacenc_is.h"
+#include "aacenc_tns.h"
/** Frequency in Hz for lower limit of noise substitution **/
#define NOISE_LOW_LIMIT 4500
@@ -958,8 +959,10 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
search_for_quantizers_faac,
encode_window_bands_info,
quantize_and_encode_band,
+ encode_tns_info,
set_special_band_scalefactors,
search_for_pns,
+ search_for_tns,
search_for_ms,
search_for_is,
},
@@ -967,8 +970,10 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
search_for_quantizers_anmr,
encode_window_bands_info,
quantize_and_encode_band,
+ encode_tns_info,
set_special_band_scalefactors,
search_for_pns,
+ search_for_tns,
search_for_ms,
search_for_is,
},
@@ -976,8 +981,10 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
search_for_quantizers_twoloop,
codebook_trellis_rate,
quantize_and_encode_band,
+ encode_tns_info,
set_special_band_scalefactors,
search_for_pns,
+ search_for_tns,
search_for_ms,
search_for_is,
},
@@ -985,8 +992,10 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
search_for_quantizers_fast,
encode_window_bands_info,
quantize_and_encode_band,
+ encode_tns_info,
set_special_band_scalefactors,
search_for_pns,
+ search_for_tns,
search_for_ms,
search_for_is,
},
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 2775dd9f7a..5b05b993c4 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -401,7 +401,10 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
encode_pulses(s, &sce->pulse);
- put_bits(&s->pb, 1, 0); //tns
+ if (s->coder->encode_tns_info)
+ s->coder->encode_tns_info(s, sce);
+ else
+ put_bits(&s->pb, 1, 0);
put_bits(&s->pb, 1, 0); //ssr
encode_spectral_coeffs(s, sce);
return 0;
@@ -571,6 +574,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
for (ch = 0; ch < chans; ch++) {
sce = &cpe->ch[ch];
coeffs[ch] = sce->coeffs;
+ memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
for (w = 0; w < 128; w++)
if (sce->band_type[w] > RESERVED_BT)
sce->band_type[w] = 0;
@@ -593,11 +597,15 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
}
}
- if (s->options.pns && s->coder->search_for_pns) {
- for (ch = 0; ch < chans; ch++) {
- s->cur_channel = start_ch + ch;
- s->coder->search_for_pns(s, avctx, &cpe->ch[ch]);
- }
+ for (ch = 0; ch < chans; ch++) {
+ 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 (sce->tns.present)
+ tns_mode = 1;
}
s->cur_channel = start_ch;
if (s->options.stereo_mode && cpe->common_window) {
@@ -824,6 +832,9 @@ static const AVOption aacenc_options[] = {
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 0}, 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"},
{NULL}
};
diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
index eb3e378b76..240be7d1a7 100644
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -45,6 +45,7 @@ typedef struct AACEncOptions {
int stereo_mode;
int aac_coder;
int pns;
+ int tns;
int pred;
int intensity_stereo;
} AACEncOptions;
@@ -58,8 +59,10 @@ typedef struct AACCoefficientsEncoder {
int win, int group_len, const float lambda);
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 (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
+ void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
} AACCoefficientsEncoder;
diff --git a/libavcodec/aacenc_tns.c b/libavcodec/aacenc_tns.c
new file mode 100644
index 0000000000..50ac29c05c
--- /dev/null
+++ b/libavcodec/aacenc_tns.c
@@ -0,0 +1,235 @@
+/*
+ * 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 temporal noise shaping
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include <strings.h>
+#include "aacenc.h"
+#include "aacenc_tns.h"
+#include "aactab.h"
+#include "aacenc_utils.h"
+#include "aacenc_quantization.h"
+
+static inline void conv_to_int32(int32_t *loc, float *samples, int num, float norm)
+{
+ int i;
+ for (i = 0; i < num; i++)
+ loc[i] = ceilf((samples[i]/norm)*INT32_MAX);
+}
+
+static inline void conv_to_float(float *arr, int32_t *cof, int num)
+{
+ int i;
+ for (i = 0; i < num; i++)
+ arr[i] = (float)cof[i]/INT32_MAX;
+}
+
+/* Input: quantized 4 bit coef, output: 1 if first (MSB) 2 bits are the same */
+static inline int coef_test_compression(int coef)
+{
+ int res = 0;
+ coef = coef >> 3;
+ res += ffs(coef);
+ coef = coef >> 1;
+ res += ffs(coef);
+ return res == 1 ? 0 : 1;
+}
+
+static inline int compress_coef(int *coefs, int num)
+{
+ int i, res = 0;
+ for (i = 0; i < num; i++)
+ res += coef_test_compression(coefs[i]);
+ return res != num ? 0 : 1;
+}
+
+/**
+ * Encode TNS data.
+ * Coefficient compression saves a single bit.
+ */
+void encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
+{
+ int i, w, filt, coef_len, coef_compress;
+ const int coef_res = MAX_LPC_PRECISION == 4 ? 1 : 0;
+ const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+
+ put_bits(&s->pb, 1, !!sce->tns.present);
+
+ if (!sce->tns.present)
+ return;
+
+ for (i = 0; i < sce->ics.num_windows; i++) {
+ put_bits(&s->pb, 2 - is8, sce->tns.n_filt[i]);
+ if (sce->tns.n_filt[i]) {
+ put_bits(&s->pb, 1, !!coef_res);
+ for (filt = 0; filt < sce->tns.n_filt[i]; filt++) {
+ put_bits(&s->pb, 6 - 2 * is8, sce->tns.length[i][filt]);
+ put_bits(&s->pb, 5 - 2 * is8, sce->tns.order[i][filt]);
+ if (sce->tns.order[i][filt]) {
+ coef_compress = compress_coef(sce->tns.coef_idx[i][filt],
+ sce->tns.order[i][filt]);
+ put_bits(&s->pb, 1, !!sce->tns.direction[i][filt]);
+ put_bits(&s->pb, 1, !!coef_compress);
+ coef_len = coef_res + 3 - coef_compress;
+ for (w = 0; w < sce->tns.order[i][filt]; w++)
+ put_bits(&s->pb, coef_len, sce->tns.coef_idx[i][filt][w]);
+ }
+ }
+ }
+ }
+}
+
+static int process_tns_coeffs(TemporalNoiseShaping *tns, float *tns_coefs_raw,
+ int order, int w, int filt)
+{
+ int i, j;
+ int *idx = tns->coef_idx[w][filt];
+ float *lpc = tns->coef[w][filt];
+ const int iqfac_p = ((1 << (MAX_LPC_PRECISION-1)) - 0.5)/(M_PI/2.0);
+ const int iqfac_m = ((1 << (MAX_LPC_PRECISION-1)) + 0.5)/(M_PI/2.0);
+ float temp[TNS_MAX_ORDER] = {0.0f}, out[TNS_MAX_ORDER] = {0.0f};
+
+ /* Quantization */
+ for (i = 0; i < order; i++) {
+ idx[i] = ceilf(asin(tns_coefs_raw[i])*((tns_coefs_raw[i] >= 0) ? iqfac_p : iqfac_m));
+ lpc[i] = 2*sin(idx[i]/((idx[i] >= 0) ? iqfac_p : iqfac_m));
+ }
+
+ /* Trim any coeff less than 0.1f from the end */
+ for (i = order; i > -1; i--) {
+ lpc[i] = (fabs(lpc[i]) > 0.1f) ? lpc[i] : 0.0f;
+ if (lpc[i] != 0.0 ) {
+ order = i;
+ break;
+ }
+ }
+
+ if (!order)
+ return 0;
+
+ /* Step up procedure, convert to LPC coeffs */
+ out[0] = 1.0f;
+ for (i = 1; i <= order; i++) {
+ for (j = 1; j < i; j++) {
+ temp[j] = out[j] + lpc[i]*out[i-j];
+ }
+ for (j = 1; j <= i; j++) {
+ out[j] = temp[j];
+ }
+ out[i] = lpc[i-1];
+ }
+ memcpy(lpc, out, TNS_MAX_ORDER*sizeof(float));
+
+ return order;
+}
+
+static void apply_tns_filter(float *out, float *in, int order, int direction,
+ float *tns_coefs, int ltp_used, int w, int filt, int start_i, int len)
+{
+ int i, j, inc, start = start_i;
+ float tmp[TNS_MAX_ORDER+1];
+ if (direction) {
+ inc = -1;
+ start = (start + len) - 1;
+ } else {
+ inc = 1;
+ }
+ if (!ltp_used) { /* AR filter */
+ for (i = 0; i < len; i++, start += inc)
+ out[i] = in[start];
+ for (j = 1; j <= FFMIN(i, order); j++)
+ out[i] += tns_coefs[j]*in[start - j*inc];
+ } else { /* MA filter */
+ for (i = 0; i < len; i++, start += inc) {
+ tmp[0] = out[i] = in[start];
+ for (j = 1; j <= FFMIN(i, order); j++)
+ out[i] += tmp[j]*tns_coefs[j];
+ for (j = order; j > 0; j--)
+ tmp[j] = tmp[j - 1];
+ }
+ }
+}
+
+void search_for_tns(AACEncContext *s, SingleChannelElement *sce)
+{
+ TemporalNoiseShaping *tns = &sce->tns;
+ int w, g, order, sfb_start, sfb_len, coef_start, shift[MAX_LPC_ORDER], count = 0;
+ const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+ const int tns_max_order = is8 ? 7 : s->profile == FF_PROFILE_AAC_LOW ? 12 : TNS_MAX_ORDER;
+ const float freq_mult = mpeg4audio_sample_rates[s->samplerate_index]/(1024.0f/sce->ics.num_windows)/2.0f;
+ float max_coef = 0.0f;
+
+ for (coef_start = 0; coef_start < 1024; coef_start++)
+ max_coef = FFMAX(max_coef, sce->pcoeffs[coef_start]);
+
+ for (w = 0; w < sce->ics.num_windows; w++) {
+ int filters = 1, start = 0, coef_len = 0;
+ int32_t conv_coeff[1024] = {0};
+ int32_t coefs_t[MAX_LPC_ORDER][MAX_LPC_ORDER] = {{0}};
+
+ /* Determine start sfb + coef - excludes anything below threshold */
+ for (g = 0; g < sce->ics.num_swb; g++) {
+ if (start*freq_mult > TNS_LOW_LIMIT) {
+ sfb_start = w*16+g;
+ sfb_len = (w+1)*16 + g - sfb_start;
+ coef_start = sce->ics.swb_offset[sfb_start];
+ coef_len = sce->ics.swb_offset[sfb_start + sfb_len] - coef_start;
+ break;
+ }
+ start += sce->ics.swb_sizes[g];
+ }
+
+ if (coef_len <= 0)
+ continue;
+
+ conv_to_int32(conv_coeff, &sce->pcoeffs[coef_start], coef_len, max_coef);
+
+ /* LPC */
+ order = ff_lpc_calc_coefs(&s->lpc, conv_coeff, coef_len,
+ TNS_MIN_PRED_ORDER, tns_max_order,
+ 32, coefs_t, shift,
+ FF_LPC_TYPE_LEVINSON, 10,
+ ORDER_METHOD_EST, MAX_LPC_SHIFT, 0) - 1;
+
+ /* Works surprisingly well, remember to tweak MAX_LPC_SHIFT if you want to play around with this */
+ if (shift[order] > 3) {
+ int direction = 0;
+ float tns_coefs_raw[TNS_MAX_ORDER];
+ tns->n_filt[w] = filters++;
+ conv_to_float(tns_coefs_raw, coefs_t[order], order);
+ for (g = 0; g < tns->n_filt[w]; g++) {
+ process_tns_coeffs(tns, tns_coefs_raw, order, w, g);
+ apply_tns_filter(&sce->coeffs[coef_start], sce->pcoeffs, order, direction, tns->coef[w][g],
+ sce->ics.ltp.present, w, g, coef_start, coef_len);
+ tns->order[w][g] = order;
+ tns->length[w][g] = sfb_len;
+ tns->direction[w][g] = direction;
+ }
+ count++;
+ }
+ }
+
+ sce->tns.present = !!count;
+}
diff --git a/libavcodec/aacenc_tns.h b/libavcodec/aacenc_tns.h
new file mode 100644
index 0000000000..3b11ca03b9
--- /dev/null
+++ b/libavcodec/aacenc_tns.h
@@ -0,0 +1,45 @@
+/*
+ * 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 temporal noise shaping
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_TNS_H
+#define AVCODEC_AACENC_TNS_H
+
+#include "aac.h"
+
+/** Frequency in Hz for lower limit of TNS **/
+#define TNS_LOW_LIMIT 2150
+
+/** LPC settings */
+#define TNS_MIN_PRED_ORDER 0
+#define MAX_LPC_PRECISION 4 /* 4 bits ltp coeff precision */
+#define TNS_LPC_PASSES 2
+#define MAX_LPC_SHIFT 4
+
+void encode_tns_info(AACEncContext *s, SingleChannelElement *sce);
+void search_for_tns(AACEncContext *s, SingleChannelElement *sce);
+
+#endif /* AVCODEC_AACENC_TNS_H */
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-10 22:28:53 +0000