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aacenc_ltp.c
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1 /*
2  * AAC encoder long term prediction extension
3  * Copyright (C) 2015 Rostislav Pehlivanov
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * AAC encoder long term prediction extension
25  * @author Rostislav Pehlivanov ( atomnuker gmail com )
26  */
27 
28 #include "aacenc_ltp.h"
29 #include "aacenc_quantization.h"
30 #include "aacenc_utils.h"
31 
32 /**
33  * Encode LTP data.
34  */
36  int common_window)
37 {
38  int i;
39  IndividualChannelStream *ics = &sce->ics;
40  if (s->profile != FF_PROFILE_AAC_LTP || !ics->predictor_present)
41  return;
42  if (common_window)
43  put_bits(&s->pb, 1, 0);
44  put_bits(&s->pb, 1, ics->ltp.present);
45  if (!ics->ltp.present)
46  return;
47  put_bits(&s->pb, 11, ics->ltp.lag);
48  put_bits(&s->pb, 3, ics->ltp.coef_idx);
49  for (i = 0; i < FFMIN(ics->max_sfb, MAX_LTP_LONG_SFB); i++)
50  put_bits(&s->pb, 1, ics->ltp.used[i]);
51 }
52 
54 {
55  int i, ch, tag, chans, cur_channel, start_ch = 0;
56  ChannelElement *cpe;
58  for (i = 0; i < s->chan_map[0]; i++) {
59  cpe = &s->cpe[i];
60  tag = s->chan_map[i+1];
61  chans = tag == TYPE_CPE ? 2 : 1;
62  for (ch = 0; ch < chans; ch++) {
63  sce = &cpe->ch[ch];
64  cur_channel = start_ch + ch;
65  /* New sample + overlap */
66  memcpy(&sce->ltp_state[0], &sce->ltp_state[1024], 1024*sizeof(sce->ltp_state[0]));
67  memcpy(&sce->ltp_state[1024], &s->planar_samples[cur_channel][2048], 1024*sizeof(sce->ltp_state[0]));
68  memcpy(&sce->ltp_state[2048], &sce->ret_buf[0], 1024*sizeof(sce->ltp_state[0]));
69  sce->ics.ltp.lag = 0;
70  }
71  start_ch += chans;
72  }
73 }
74 
75 static void get_lag(float *buf, const float *new, LongTermPrediction *ltp)
76 {
77  int i, j, lag = 0, max_corr = 0;
78  float max_ratio = 0.0f;
79  for (i = 0; i < 2048; i++) {
80  float corr, s0 = 0.0f, s1 = 0.0f;
81  const int start = FFMAX(0, i - 1024);
82  for (j = start; j < 2048; j++) {
83  const int idx = j - i + 1024;
84  s0 += new[j]*buf[idx];
85  s1 += buf[idx]*buf[idx];
86  }
87  corr = s1 > 0.0f ? s0/sqrt(s1) : 0.0f;
88  if (corr > max_corr) {
89  max_corr = corr;
90  lag = i;
91  max_ratio = corr/(2048-start);
92  }
93  }
94  ltp->lag = FFMAX(av_clip_uintp2(lag, 11), 0);
95  ltp->coef_idx = quant_array_idx(max_ratio, ltp_coef, 8);
96  ltp->coef = ltp_coef[ltp->coef_idx];
97 }
98 
99 static void generate_samples(float *buf, LongTermPrediction *ltp)
100 {
101  int i, samples_num = 2048;
102  if (!ltp->lag) {
103  ltp->present = 0;
104  return;
105  } else if (ltp->lag < 1024) {
106  samples_num = ltp->lag + 1024;
107  }
108  for (i = 0; i < samples_num; i++)
109  buf[i] = ltp->coef*buf[i + 2048 - ltp->lag];
110  memset(&buf[i], 0, (2048 - i)*sizeof(float));
111 }
112 
113 /**
114  * Process LTP parameters
115  * @see Patent WO2006070265A1
116  */
118 {
119  float *pred_signal = &sce->ltp_state[0];
120  const float *samples = &s->planar_samples[s->cur_channel][1024];
121 
122  if (s->profile != FF_PROFILE_AAC_LTP)
123  return;
124 
125  /* Calculate lag */
126  get_lag(pred_signal, samples, &sce->ics.ltp);
127  generate_samples(pred_signal, &sce->ics.ltp);
128 }
129 
131 {
132  int sfb, count = 0;
133  SingleChannelElement *sce0 = &cpe->ch[0];
134  SingleChannelElement *sce1 = &cpe->ch[1];
135 
136  if (!cpe->common_window ||
139  sce0->ics.ltp.present = 0;
140  return;
141  }
142 
143  for (sfb = 0; sfb < FFMIN(sce0->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++) {
144  int sum = sce0->ics.ltp.used[sfb] + sce1->ics.ltp.used[sfb];
145  if (sum != 2) {
146  sce0->ics.ltp.used[sfb] = 0;
147  } else if (sum == 2) {
148  count++;
149  }
150  }
151 
152  sce0->ics.ltp.present = !!count;
153  sce0->ics.predictor_present = !!count;
154 }
155 
156 /**
157  * Mark LTP sfb's
158  */
160  int common_window)
161 {
162  int w, g, w2, i, start = 0, count = 0;
163  int saved_bits = -(15 + FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB));
164  float *C34 = &s->scoefs[128*0], *PCD = &s->scoefs[128*1];
165  float *PCD34 = &s->scoefs[128*2];
166  const int max_ltp = FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB);
167 
168  if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
169  if (sce->ics.ltp.lag) {
170  memset(&sce->ltp_state[0], 0, 3072*sizeof(sce->ltp_state[0]));
171  memset(&sce->ics.ltp, 0, sizeof(LongTermPrediction));
172  }
173  return;
174  }
175 
176  if (!sce->ics.ltp.lag || s->lambda > 120.0f)
177  return;
178 
179  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
180  start = 0;
181  for (g = 0; g < sce->ics.num_swb; g++) {
182  int bits1 = 0, bits2 = 0;
183  float dist1 = 0.0f, dist2 = 0.0f;
184  if (w*16+g > max_ltp) {
185  start += sce->ics.swb_sizes[g];
186  continue;
187  }
188  for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
189  int bits_tmp1, bits_tmp2;
190  FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
191  for (i = 0; i < sce->ics.swb_sizes[g]; i++)
192  PCD[i] = sce->coeffs[start+(w+w2)*128+i] - sce->lcoeffs[start+(w+w2)*128+i];
193  s->abs_pow34(C34, &sce->coeffs[start+(w+w2)*128], sce->ics.swb_sizes[g]);
194  s->abs_pow34(PCD34, PCD, sce->ics.swb_sizes[g]);
195  dist1 += quantize_band_cost(s, &sce->coeffs[start+(w+w2)*128], C34, sce->ics.swb_sizes[g],
196  sce->sf_idx[(w+w2)*16+g], sce->band_type[(w+w2)*16+g],
197  s->lambda/band->threshold, INFINITY, &bits_tmp1, NULL, 0);
198  dist2 += quantize_band_cost(s, PCD, PCD34, sce->ics.swb_sizes[g],
199  sce->sf_idx[(w+w2)*16+g],
200  sce->band_type[(w+w2)*16+g],
201  s->lambda/band->threshold, INFINITY, &bits_tmp2, NULL, 0);
202  bits1 += bits_tmp1;
203  bits2 += bits_tmp2;
204  }
205  if (dist2 < dist1 && bits2 < bits1) {
206  for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
207  for (i = 0; i < sce->ics.swb_sizes[g]; i++)
208  sce->coeffs[start+(w+w2)*128+i] -= sce->lcoeffs[start+(w+w2)*128+i];
209  sce->ics.ltp.used[w*16+g] = 1;
210  saved_bits += bits1 - bits2;
211  count++;
212  }
213  start += sce->ics.swb_sizes[g];
214  }
215  }
216 
217  sce->ics.ltp.present = !!count && (saved_bits >= 0);
218  sce->ics.predictor_present = !!sce->ics.ltp.present;
219 
220  /* Reset any marked sfbs */
221  if (!sce->ics.ltp.present && !!count) {
222  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
223  start = 0;
224  for (g = 0; g < sce->ics.num_swb; g++) {
225  if (sce->ics.ltp.used[w*16+g]) {
226  for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
227  for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
228  sce->coeffs[start+(w+w2)*128+i] += sce->lcoeffs[start+(w+w2)*128+i];
229  }
230  }
231  }
232  start += sce->ics.swb_sizes[g];
233  }
234  }
235  }
236 }
AAC encoder long term prediction extension.
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:768
void ff_aac_encode_ltp_info(AACEncContext *s, SingleChannelElement *sce, int common_window)
Encode LTP data.
Definition: aacenc_ltp.c:35
void ff_aac_ltp_insert_new_frame(AACEncContext *s)
Definition: aacenc_ltp.c:53
float dist2
Definition: aacenc_is.h:41
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:206
const char * g
Definition: vf_curves.c:112
FFPsyBand psy_bands[PSY_MAX_BANDS]
channel bands information
Definition: psymodel.h:61
Definition: aac.h:57
int common_window
Set if channels share a common 'IndividualChannelStream' in bitstream.
Definition: aac.h:278
float lambda
Definition: aacenc.h:116
#define MAX_LTP_LONG_SFB
Definition: aac.h:51
AAC encoder quantizer.
AAC encoder context.
Definition: aacenc.h:95
SingleChannelElement ch[2]
Definition: aac.h:284
const uint8_t * chan_map
channel configuration map
Definition: aacenc.h:108
#define FF_PROFILE_AAC_LTP
Definition: avcodec.h:3242
static const uint8_t bits2[81]
Definition: aactab.c:130
uint32_t tag
Definition: movenc.c:1413
single band psychoacoustic information
Definition: psymodel.h:50
int profile
copied from avctx
Definition: aacenc.h:104
float * planar_samples[8]
saved preprocessed input
Definition: aacenc.h:102
void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce)
Process LTP parameters.
Definition: aacenc_ltp.c:117
float dist1
Definition: aacenc_is.h:40
#define s0
Definition: regdef.h:37
static void generate_samples(float *buf, LongTermPrediction *ltp)
Definition: aacenc_ltp.c:99
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
GLsizei count
Definition: opengl_enc.c:109
int num_swb
number of scalefactor window bands
Definition: aac.h:183
#define FFMAX(a, b)
Definition: common.h:94
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:264
void(* abs_pow34)(float *out, const float *in, const int size)
Definition: aacenc.h:129
enum WindowSequence window_sequence[2]
Definition: aac.h:176
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:265
int cur_channel
current channel for coder context
Definition: aacenc.h:114
#define FFMIN(a, b)
Definition: common.h:96
static const INTFLOAT ltp_coef[8]
Definition: aactab.h:94
PutBitContext pb
Definition: aacenc.h:98
#define INFINITY
Definition: math.h:27
static int quant_array_idx(const float val, const float *arr, const int num)
Definition: aacenc_utils.h:171
Long Term Prediction.
Definition: aac.h:163
IndividualChannelStream ics
Definition: aac.h:249
void * buf
Definition: avisynth_c.h:690
uint8_t group_len[8]
Definition: aac.h:179
const uint8_t * swb_sizes
table of scalefactor band sizes for a particular window
Definition: aac.h:182
#define s1
Definition: regdef.h:38
FFPsyContext psy
Definition: aacenc.h:111
LongTermPrediction ltp
Definition: aac.h:180
int sf_idx[128]
scalefactor indices (used by encoder)
Definition: aac.h:256
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:262
int16_t lag
Definition: aac.h:165
AAC encoder utilities.
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:248
ChannelElement * cpe
channel elements
Definition: aacenc.h:110
Individual Channel Stream.
Definition: aac.h:174
INTFLOAT coef
Definition: aac.h:167
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:275
AAC_FLOAT lcoeffs[1024]
MDCT of LTP coefficients (used by encoder)
Definition: aac.h:266
FFPsyChannel * ch
single channel information
Definition: psymodel.h:93
enum BandType band_type[128]
band types
Definition: aac.h:252
static void get_lag(float *buf, const float *new, LongTermPrediction *ltp)
Definition: aacenc_ltp.c:75
void ff_aac_adjust_common_ltp(AACEncContext *s, ChannelElement *cpe)
Definition: aacenc_ltp.c:130
void ff_aac_search_for_ltp(AACEncContext *s, SingleChannelElement *sce, int common_window)
Mark LTP sfb's.
Definition: aacenc_ltp.c:159
void INT64 start
Definition: avisynth_c.h:690
static float quantize_band_cost(struct AACEncContext *s, const float *in, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy, int rtz)
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:168
int8_t present
Definition: aac.h:164
float threshold
Definition: psymodel.h:53
float scoefs[1024]
scaled coefficients
Definition: aacenc.h:124
static const uint8_t bits1[81]
Definition: aactab.c:107