FFmpeg
afir_template.c
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1 /*
2  * Copyright (c) 2017 Paul B Mahol
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/tx.h"
22 #include "avfilter.h"
23 #include "audio.h"
24 
25 #undef ctype
26 #undef ftype
27 #undef SQRT
28 #undef HYPOT
29 #undef SAMPLE_FORMAT
30 #undef TX_TYPE
31 #undef FABS
32 #undef POW
33 #if DEPTH == 32
34 #define SAMPLE_FORMAT float
35 #define SQRT sqrtf
36 #define HYPOT hypotf
37 #define ctype AVComplexFloat
38 #define ftype float
39 #define TX_TYPE AV_TX_FLOAT_RDFT
40 #define FABS fabsf
41 #define POW powf
42 #else
43 #define SAMPLE_FORMAT double
44 #define SQRT sqrt
45 #define HYPOT hypot
46 #define ctype AVComplexDouble
47 #define ftype double
48 #define TX_TYPE AV_TX_DOUBLE_RDFT
49 #define FABS fabs
50 #define POW pow
51 #endif
52 
53 #define fn3(a,b) a##_##b
54 #define fn2(a,b) fn3(a,b)
55 #define fn(a) fn2(a, SAMPLE_FORMAT)
56 
57 static ftype fn(ir_gain)(AVFilterContext *ctx, AudioFIRContext *s,
58  int cur_nb_taps, const ftype *time)
59 {
60  ftype ch_gain, sum = 0;
61 
62  if (s->ir_norm < 0.f) {
63  ch_gain = 1;
64  } else if (s->ir_norm == 0.f) {
65  for (int i = 0; i < cur_nb_taps; i++)
66  sum += time[i];
67  ch_gain = 1. / sum;
68  } else {
69  ftype ir_norm = s->ir_norm;
70 
71  for (int i = 0; i < cur_nb_taps; i++)
72  sum += POW(FABS(time[i]), ir_norm);
73  ch_gain = 1. / POW(sum, 1. / ir_norm);
74  }
75 
76  return ch_gain;
77 }
78 
79 static void fn(ir_scale)(AVFilterContext *ctx, AudioFIRContext *s,
80  int cur_nb_taps, int ch,
81  ftype *time, ftype ch_gain)
82 {
83  if (ch_gain != 1. || s->ir_gain != 1.) {
84  ftype gain = ch_gain * s->ir_gain;
85 
86  av_log(ctx, AV_LOG_DEBUG, "ch%d gain %f\n", ch, gain);
87 #if DEPTH == 32
88  s->fdsp->vector_fmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 4));
89 #else
90  s->fdsp->vector_dmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 8));
91 #endif
92  }
93 }
94 
95 static void fn(convert_channel)(AVFilterContext *ctx, AudioFIRContext *s, int ch,
96  AudioFIRSegment *seg, int coeff_partition, int selir)
97 {
98  const int coffset = coeff_partition * seg->coeff_size;
99  const int nb_taps = s->nb_taps[selir];
100  ftype *time = (ftype *)s->norm_ir[selir]->extended_data[ch];
101  ftype *tempin = (ftype *)seg->tempin->extended_data[ch];
102  ftype *tempout = (ftype *)seg->tempout->extended_data[ch];
103  ctype *coeff = (ctype *)seg->coeff->extended_data[ch];
104  const int remaining = nb_taps - (seg->input_offset + coeff_partition * seg->part_size);
105  const int size = remaining >= seg->part_size ? seg->part_size : remaining;
106 
107  memset(tempin + size, 0, sizeof(*tempin) * (seg->block_size - size));
108  memcpy(tempin, time + seg->input_offset + coeff_partition * seg->part_size,
109  size * sizeof(*tempin));
110  seg->ctx_fn(seg->ctx[ch], tempout, tempin, sizeof(*tempin));
111  memcpy(coeff + coffset, tempout, seg->coeff_size * sizeof(*coeff));
112 
113  av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
114  av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
115  av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
116  av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
117  av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
118  av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
119  av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
120  av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
121 }
122 
123 static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples)
124 {
125  if ((nb_samples & 15) == 0 && nb_samples >= 8) {
126 #if DEPTH == 32
127  s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
128 #else
129  s->fdsp->vector_dmac_scalar(dst, src, 1.0, nb_samples);
130 #endif
131  } else {
132  for (int n = 0; n < nb_samples; n++)
133  dst[n] += src[n];
134  }
135 }
136 
137 static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int ioffset, int offset, int selir)
138 {
139  AudioFIRContext *s = ctx->priv;
140  const ftype *in = (const ftype *)s->in->extended_data[ch] + ioffset;
141  ftype *blockout, *ptr = (ftype *)out->extended_data[ch] + offset;
142  const int min_part_size = s->min_part_size;
143  const int nb_samples = FFMIN(min_part_size, out->nb_samples - offset);
144  const int nb_segments = s->nb_segments[selir];
145  const float dry_gain = s->dry_gain;
146  const float wet_gain = s->wet_gain;
147 
148  for (int segment = 0; segment < nb_segments; segment++) {
149  AudioFIRSegment *seg = &s->seg[selir][segment];
150  ftype *src = (ftype *)seg->input->extended_data[ch];
151  ftype *dst = (ftype *)seg->output->extended_data[ch];
152  ftype *sumin = (ftype *)seg->sumin->extended_data[ch];
153  ftype *sumout = (ftype *)seg->sumout->extended_data[ch];
154  ftype *tempin = (ftype *)seg->tempin->extended_data[ch];
155  ftype *buf = (ftype *)seg->buffer->extended_data[ch];
156  int *output_offset = &seg->output_offset[ch];
157  const int nb_partitions = seg->nb_partitions;
158  const int input_offset = seg->input_offset;
159  const int part_size = seg->part_size;
160  int j;
161 
162  seg->part_index[ch] = seg->part_index[ch] % nb_partitions;
163  if (dry_gain == 1.f) {
164  memcpy(src + input_offset, in, nb_samples * sizeof(*src));
165  } else if (min_part_size >= 8) {
166 #if DEPTH == 32
167  s->fdsp->vector_fmul_scalar(src + input_offset, in, dry_gain, FFALIGN(nb_samples, 4));
168 #else
169  s->fdsp->vector_dmul_scalar(src + input_offset, in, dry_gain, FFALIGN(nb_samples, 8));
170 #endif
171  } else {
172  ftype *src2 = src + input_offset;
173  for (int n = 0; n < nb_samples; n++)
174  src2[n] = in[n] * dry_gain;
175  }
176 
177  output_offset[0] += min_part_size;
178  if (output_offset[0] >= part_size) {
179  output_offset[0] = 0;
180  } else {
181  memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src));
182 
183  dst += output_offset[0];
184  fn(fir_fadd)(s, ptr, dst, nb_samples);
185  continue;
186  }
187 
188  memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
189 
190  blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
191  memset(tempin + part_size, 0, sizeof(*tempin) * (seg->block_size - part_size));
192  memcpy(tempin, src, sizeof(*src) * part_size);
193  seg->tx_fn(seg->tx[ch], blockout, tempin, sizeof(ftype));
194 
195  j = seg->part_index[ch];
196  for (int i = 0; i < nb_partitions; i++) {
197  const int input_partition = j;
198  const int coeff_partition = i;
199  const int coffset = coeff_partition * seg->coeff_size;
200  const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + input_partition * seg->block_size;
201  const ctype *coeff = ((const ctype *)seg->coeff->extended_data[ch]) + coffset;
202 
203  if (j == 0)
204  j = nb_partitions;
205  j--;
206 
207 #if DEPTH == 32
208  s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, part_size);
209 #else
210  s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, part_size);
211 #endif
212  }
213 
214  seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ctype));
215 
216  fn(fir_fadd)(s, buf, sumout, part_size);
217  memcpy(dst, buf, part_size * sizeof(*dst));
218  memcpy(buf, sumout + part_size, part_size * sizeof(*buf));
219 
220  fn(fir_fadd)(s, ptr, dst, nb_samples);
221 
222  if (part_size != min_part_size)
223  memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src));
224 
225  seg->part_index[ch] = (seg->part_index[ch] + 1) % nb_partitions;
226  }
227 
228  if (wet_gain == 1.f)
229  return 0;
230 
231  if (min_part_size >= 8) {
232 #if DEPTH == 32
233  s->fdsp->vector_fmul_scalar(ptr, ptr, wet_gain, FFALIGN(nb_samples, 4));
234 #else
235  s->fdsp->vector_dmul_scalar(ptr, ptr, wet_gain, FFALIGN(nb_samples, 8));
236 #endif
237  } else {
238  for (int n = 0; n < nb_samples; n++)
239  ptr[n] *= wet_gain;
240  }
241 
242  return 0;
243 }
244 
245 static void fn(fir_quantums)(AVFilterContext *ctx, AudioFIRContext *s, AVFrame *out,
246  int min_part_size, int ch, int offset,
247  int prev_selir, int selir)
248 {
249  if (ctx->is_disabled || s->prev_is_disabled) {
250  const ftype *in = (const ftype *)s->in->extended_data[ch] + offset;
251  const ftype *xfade0 = (const ftype *)s->xfade[0]->extended_data[ch];
252  const ftype *xfade1 = (const ftype *)s->xfade[1]->extended_data[ch];
253  ftype *src0 = (ftype *)s->fadein[0]->extended_data[ch];
254  ftype *src1 = (ftype *)s->fadein[1]->extended_data[ch];
255  ftype *dst = ((ftype *)out->extended_data[ch]) + offset;
256 
257  if (ctx->is_disabled && !s->prev_is_disabled) {
258  memset(src0, 0, min_part_size * sizeof(ftype));
259  fn(fir_quantum)(ctx, s->fadein[0], ch, offset, 0, selir);
260  for (int n = 0; n < min_part_size; n++)
261  dst[n] = xfade1[n] * src0[n] + xfade0[n] * in[n];
262  } else if (!ctx->is_disabled && s->prev_is_disabled) {
263  memset(src1, 0, min_part_size * sizeof(ftype));
264  fn(fir_quantum)(ctx, s->fadein[1], ch, offset, 0, selir);
265  for (int n = 0; n < min_part_size; n++)
266  dst[n] = xfade1[n] * in[n] + xfade0[n] * src1[n];
267  } else {
268  memcpy(dst, in, sizeof(ftype) * min_part_size);
269  }
270  } else if (prev_selir != selir && s->loading[ch] != 0) {
271  const ftype *xfade0 = (const ftype *)s->xfade[0]->extended_data[ch];
272  const ftype *xfade1 = (const ftype *)s->xfade[1]->extended_data[ch];
273  ftype *src0 = (ftype *)s->fadein[0]->extended_data[ch];
274  ftype *src1 = (ftype *)s->fadein[1]->extended_data[ch];
275  ftype *dst = ((ftype *)out->extended_data[ch]) + offset;
276 
277  memset(src0, 0, min_part_size * sizeof(ftype));
278  memset(src1, 0, min_part_size * sizeof(ftype));
279 
280  fn(fir_quantum)(ctx, s->fadein[0], ch, offset, 0, prev_selir);
281  fn(fir_quantum)(ctx, s->fadein[1], ch, offset, 0, selir);
282 
283  if (s->loading[ch] > s->max_offset[selir]) {
284  for (int n = 0; n < min_part_size; n++)
285  dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n];
286  s->loading[ch] = 0;
287  } else {
288  memcpy(dst, src0, min_part_size * sizeof(ftype));
289  }
290  } else {
291  fn(fir_quantum)(ctx, out, ch, offset, offset, selir);
292  }
293 }
convert_channel
static void fn() convert_channel(AVFilterContext *ctx, AudioFIRContext *s, int ch, AudioFIRSegment *seg, int coeff_partition, int selir)
Definition: afir_template.c:95
AudioFIRSegment::block_size
int block_size
Definition: af_afir.c:51
out
FILE * out
Definition: movenc.c:55
ctype
#define ctype
Definition: afir_template.c:46
src1
const pixel * src1
Definition: h264pred_template.c:421
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
AudioFIRSegment::buffer
AVFrame * buffer
Definition: af_afir.c:65
AudioFIRSegment::input_offset
int input_offset
Definition: af_afir.c:55
AudioFIRSegment::tx_fn
av_tx_fn tx_fn
Definition: af_afir.c:71
AudioFIRSegment::part_size
int part_size
Definition: af_afir.c:50
AudioFIRSegment::input_size
int input_size
Definition: af_afir.c:54
FABS
#define FABS
Definition: afir_template.c:49
AudioFIRSegment::coeff
AVFrame * coeff
Definition: af_afir.c:66
fn
#define fn(a)
Definition: afir_template.c:55
AudioFIRSegment::blockout
AVFrame * blockout
Definition: af_afir.c:62
AudioFIRSegment
Definition: af_afir.c:48
AudioFIRSegment::tx
AVTXContext ** tx
Definition: af_afir.c:70
ftype
#define ftype
Definition: afir_template.c:47
s
#define s(width, name)
Definition: cbs_vp9.c:198
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:230
ctx
AVFormatContext * ctx
Definition: movenc.c:49
AudioFIRSegment::itx_fn
av_tx_fn itx_fn
Definition: af_afir.c:71
fir_quantums
static void fn() fir_quantums(AVFilterContext *ctx, AudioFIRContext *s, AVFrame *out, int min_part_size, int ch, int offset, int prev_selir, int selir)
Definition: afir_template.c:245
AudioFIRSegment::output
AVFrame * output
Definition: af_afir.c:68
f
f
Definition: af_crystalizer.c:122
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition: dsp.h:83
size
int size
Definition: twinvq_data.h:10344
AudioFIRSegment::sumin
AVFrame * sumin
Definition: af_afir.c:60
fir_fadd
static void fn() fir_fadd(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples)
Definition: afir_template.c:123
offset
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Definition: writing_filters.txt:86
POW
#define POW
Definition: afir_template.c:50
AudioFIRSegment::tempin
AVFrame * tempin
Definition: af_afir.c:63
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
AudioFIRSegment::input
AVFrame * input
Definition: af_afir.c:67
AudioFIRSegment::coeff_size
int coeff_size
Definition: af_afir.c:53
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:450
src2
const pixel * src2
Definition: h264pred_template.c:422
AudioFIRSegment::nb_partitions
int nb_partitions
Definition: af_afir.c:49
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AudioFIRSegment::itx
AVTXContext ** itx
Definition: af_afir.c:70
ir_gain
static ftype fn() ir_gain(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps, const ftype *time)
Definition: afir_template.c:57
AudioFIRSegment::fft_length
int fft_length
Definition: af_afir.c:52
AudioFIRSegment::sumout
AVFrame * sumout
Definition: af_afir.c:61
AudioFIRContext
Definition: af_afir.c:74
avfilter.h
segment
Definition: hls.c:77
src0
const pixel *const src0
Definition: h264pred_template.c:420
AVFilterContext
An instance of a filter.
Definition: avfilter.h:457
audio.h
ir_scale
static void fn() ir_scale(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps, int ch, ftype *time, ftype ch_gain)
Definition: afir_template.c:79
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:80
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AudioFIRSegment::output_offset
int * output_offset
Definition: af_afir.c:57
src
#define src
Definition: vp8dsp.c:248
tx.h
fir_quantum
static int fn() fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int ioffset, int offset, int selir)
Definition: afir_template.c:137
AudioFIRSegment::part_index
int * part_index
Definition: af_afir.c:58
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