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avfilter/af_afir: add support for double sample format

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This commit is contained in:
Paul B Mahol 2022-05-14 10:28:49 +02:00
parent e6f0cec880
commit 163e737c17
5 changed files with 621 additions and 413 deletions
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16
doc/filters.texi
View File

@ -1655,6 +1655,22 @@ Allowed range is from @var{1} to @var{32}. Default is @var{1}.
Set IR stream which will be used for convolution, starting from @var{0}, should always be
lower than supplied value by @code{nbirs} option. Default is @var{0}.
This option can be changed at runtime via @ref{commands}.
@item precision
Set which precision to use when processing samples.
@table @option
@item auto
Auto pick internal sample format depending on other filters.
@item float
Always use single-floating point precision sample format.
@item double
Always use double-floating point precision sample format.
@end table
Default value is auto.
@end table
@subsection Examples

507
libavfilter/af_afir.c
View File

@ -42,248 +42,9 @@
#include "filters.h"
#include "formats.h"
#include "internal.h"
#include "af_afir.h"
#include "af_afirdsp.h"
typedef struct AudioFIRSegment {
int nb_partitions;
int part_size;
int block_size;
int fft_length;
int coeff_size;
int input_size;
int input_offset;
int *output_offset;
int *part_index;
AVFrame *sumin;
AVFrame *sumout;
AVFrame *blockin;
AVFrame *blockout;
AVFrame *buffer;
AVFrame *coeff;
AVFrame *input;
AVFrame *output;
AVTXContext **tx, **itx;
av_tx_fn tx_fn, itx_fn;
} AudioFIRSegment;
typedef struct AudioFIRContext {
const AVClass *class;
float wet_gain;
float dry_gain;
float length;
int gtype;
float ir_gain;
int ir_format;
float max_ir_len;
int response;
int w, h;
AVRational frame_rate;
int ir_channel;
int minp;
int maxp;
int nb_irs;
int selir;
float gain;
int eof_coeffs[32];
int have_coeffs;
int nb_taps;
int nb_channels;
int nb_coef_channels;
int one2many;
AudioFIRSegment seg[1024];
int nb_segments;
AVFrame *in;
AVFrame *ir[32];
AVFrame *video;
int min_part_size;
int64_t pts;
AudioFIRDSPContext afirdsp;
AVFloatDSPContext *fdsp;
} AudioFIRContext;
static void direct(const float *in, const AVComplexFloat *ir, int len, float *out)
{
for (int n = 0; n < len; n++)
for (int m = 0; m <= n; m++)
out[n] += ir[m].re * in[n - m];
}
static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
{
if ((nb_samples & 15) == 0 && nb_samples >= 16) {
s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
} else {
for (int n = 0; n < nb_samples; n++)
dst[n] += src[n];
}
}
static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
{
AudioFIRContext *s = ctx->priv;
const float *in = (const float *)s->in->extended_data[ch] + offset;
float *blockin, *blockout, *buf, *ptr = (float *)out->extended_data[ch] + offset;
const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
int n, i, j;
for (int segment = 0; segment < s->nb_segments; segment++) {
AudioFIRSegment *seg = &s->seg[segment];
float *src = (float *)seg->input->extended_data[ch];
float *dst = (float *)seg->output->extended_data[ch];
float *sumin = (float *)seg->sumin->extended_data[ch];
float *sumout = (float *)seg->sumout->extended_data[ch];
if (s->min_part_size >= 8) {
s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
emms_c();
} else {
for (n = 0; n < nb_samples; n++)
src[seg->input_offset + n] = in[n] * s->dry_gain;
}
seg->output_offset[ch] += s->min_part_size;
if (seg->output_offset[ch] == seg->part_size) {
seg->output_offset[ch] = 0;
} else {
memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
dst += seg->output_offset[ch];
fir_fadd(s, ptr, dst, nb_samples);
continue;
}
if (seg->part_size < 8) {
memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
j = seg->part_index[ch];
for (i = 0; i < seg->nb_partitions; i++) {
const int coffset = j * seg->coeff_size;
const AVComplexFloat *coeff = (const AVComplexFloat *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
direct(src, coeff, nb_samples, dst);
if (j == 0)
j = seg->nb_partitions;
j--;
}
seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
for (n = 0; n < nb_samples; n++) {
ptr[n] += dst[n];
}
continue;
}
memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
blockin = (float *)seg->blockin->extended_data[ch] + seg->part_index[ch] * seg->block_size;
blockout = (float *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
memset(blockin + seg->part_size, 0, sizeof(*blockin) * (seg->fft_length - seg->part_size));
memcpy(blockin, src, sizeof(*src) * seg->part_size);
seg->tx_fn(seg->tx[ch], blockout, blockin, sizeof(float));
j = seg->part_index[ch];
for (i = 0; i < seg->nb_partitions; i++) {
const int coffset = j * seg->coeff_size;
const float *blockout = (const float *)seg->blockout->extended_data[ch] + i * seg->block_size;
const AVComplexFloat *coeff = (const AVComplexFloat *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
s->afirdsp.fcmul_add(sumin, blockout, (const float *)coeff, seg->part_size);
if (j == 0)
j = seg->nb_partitions;
j--;
}
seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(float));
buf = (float *)seg->buffer->extended_data[ch];
fir_fadd(s, buf, sumout, seg->part_size);
memcpy(dst, buf, seg->part_size * sizeof(*dst));
buf = (float *)seg->buffer->extended_data[ch];
memcpy(buf, sumout + seg->part_size, seg->part_size * sizeof(*buf));
seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
fir_fadd(s, ptr, dst, nb_samples);
}
if (s->min_part_size >= 8) {
s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
emms_c();
} else {
for (n = 0; n < nb_samples; n++)
ptr[n] *= s->wet_gain;
}
return 0;
}
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
{
AudioFIRContext *s = ctx->priv;
for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
fir_quantum(ctx, out, ch, offset);
}
return 0;
}
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AVFrame *out = arg;
const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++) {
fir_channel(ctx, out, ch);
}
return 0;
}
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFrame *out = NULL;
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
out->pts = in->pts;
s->in = in;
ff_filter_execute(ctx, fir_channels, out, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
av_frame_free(&in);
s->in = NULL;
return ff_filter_frame(outlink, out);
}
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
{
const uint8_t *font;
@ -333,93 +94,64 @@ static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t col
}
}
static void draw_response(AVFilterContext *ctx, AVFrame *out)
#define DEPTH 32
#include "afir_template.c"
#undef DEPTH
#define DEPTH 64
#include "afir_template.c"
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
{
AudioFIRContext *s = ctx->priv;
float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
float min_delay = FLT_MAX, max_delay = FLT_MIN;
int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
char text[32];
int channel, i, x;
memset(out->data[0], 0, s->h * out->linesize[0]);
phase = av_malloc_array(s->w, sizeof(*phase));
mag = av_malloc_array(s->w, sizeof(*mag));
delay = av_malloc_array(s->w, sizeof(*delay));
if (!mag || !phase || !delay)
goto end;
channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->ch_layout.nb_channels - 1);
for (i = 0; i < s->w; i++) {
const float *src = (const float *)s->ir[s->selir]->extended_data[channel];
double w = i * M_PI / (s->w - 1);
double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
for (x = 0; x < s->nb_taps; x++) {
real += cos(-x * w) * src[x];
imag += sin(-x * w) * src[x];
real_num += cos(-x * w) * src[x] * x;
imag_num += sin(-x * w) * src[x] * x;
for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
fir_quantum_float(ctx, out, ch, offset);
break;
case AV_SAMPLE_FMT_DBLP:
fir_quantum_double(ctx, out, ch, offset);
break;
}
mag[i] = hypot(real, imag);
phase[i] = atan2(imag, real);
div = real * real + imag * imag;
delay[i] = (real_num * real + imag_num * imag) / div;
min = fminf(min, mag[i]);
max = fmaxf(max, mag[i]);
min_delay = fminf(min_delay, delay[i]);
max_delay = fmaxf(max_delay, delay[i]);
}
for (i = 0; i < s->w; i++) {
int ymag = mag[i] / max * (s->h - 1);
int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
return 0;
}
ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AVFrame *out = arg;
const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
if (prev_ymag < 0)
prev_ymag = ymag;
if (prev_yphase < 0)
prev_yphase = yphase;
if (prev_ydelay < 0)
prev_ydelay = ydelay;
draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
prev_ymag = ymag;
prev_yphase = yphase;
prev_ydelay = ydelay;
for (int ch = start; ch < end; ch++) {
fir_channel(ctx, out, ch);
}
if (s->w > 400 && s->h > 100) {
drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", max);
drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
return 0;
}
drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min);
drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFrame *out = NULL;
drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", max_delay);
drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min_delay);
drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
out->pts = in->pts;
end:
av_free(delay);
av_free(phase);
av_free(mag);
s->in = in;
ff_filter_execute(ctx, fir_channels, out, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
av_frame_free(&in);
s->in = NULL;
return ff_filter_frame(outlink, out);
}
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
@ -446,9 +178,20 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
return AVERROR(ENOMEM);
for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 8; ch++) {
float scale = 1.f, iscale = 1.f / part_size;
av_tx_init(&seg->tx[ch], &seg->tx_fn, AV_TX_FLOAT_RDFT, 0, 2 * part_size, &scale, 0);
av_tx_init(&seg->itx[ch], &seg->itx_fn, AV_TX_FLOAT_RDFT, 1, 2 * part_size, &iscale, 0);
double dscale = 1.0, idscale = 1.0 / part_size;
float fscale = 1.f, ifscale = 1.f / part_size;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
av_tx_init(&seg->tx[ch], &seg->tx_fn, AV_TX_FLOAT_RDFT, 0, 2 * part_size, &fscale, 0);
av_tx_init(&seg->itx[ch], &seg->itx_fn, AV_TX_FLOAT_RDFT, 1, 2 * part_size, &ifscale, 0);
break;
case AV_SAMPLE_FMT_DBLP:
av_tx_init(&seg->tx[ch], &seg->tx_fn, AV_TX_DOUBLE_RDFT, 0, 2 * part_size, &dscale, 0);
av_tx_init(&seg->itx[ch], &seg->itx_fn, AV_TX_DOUBLE_RDFT, 1, 2 * part_size, &idscale, 0);
break;
}
if (!seg->tx[ch] || !seg->itx[ch])
return AVERROR(ENOMEM);
}
@ -502,8 +245,7 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
static int convert_coeffs(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
int ret, i, ch, n, cur_nb_taps;
float power = 0;
int ret, i, cur_nb_taps;
if (!s->nb_taps) {
int part_size, max_part_size;
@ -546,109 +288,42 @@ static int convert_coeffs(AVFilterContext *ctx)
return AVERROR_BUG;
}
if (s->response)
draw_response(ctx, s->video);
if (s->response) {
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
draw_response_float(ctx, s->video);
break;
case AV_SAMPLE_FMT_DBLP:
draw_response_double(ctx, s->video);
break;
}
}
s->gain = 1;
cur_nb_taps = s->ir[s->selir]->nb_samples;
switch (s->gtype) {
case -1:
/* nothing to do */
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
ret = get_power_float(ctx, s, cur_nb_taps);
break;
case 0:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
for (i = 0; i < cur_nb_taps; i++)
power += FFABS(time[i]);
}
s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
case AV_SAMPLE_FMT_DBLP:
ret = get_power_double(ctx, s, cur_nb_taps);
break;
case 1:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
for (i = 0; i < cur_nb_taps; i++)
power += time[i];
}
s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
break;
case 2:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
for (i = 0; i < cur_nb_taps; i++)
power += time[i] * time[i];
}
s->gain = sqrtf(ch / power);
break;
default:
return AVERROR_BUG;
}
s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
}
if (ret < 0)
return ret;
av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
int toffset = 0;
for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
time[i] = 0;
av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
for (int segment = 0; segment < s->nb_segments; segment++) {
AudioFIRSegment *seg = &s->seg[segment];
float *blockin = (float *)seg->blockin->extended_data[ch];
float *blockout = (float *)seg->blockout->extended_data[ch];
AVComplexFloat *coeff = (AVComplexFloat *)seg->coeff->extended_data[ch];
av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
for (i = 0; i < seg->nb_partitions; i++) {
const int coffset = i * seg->coeff_size;
const int remaining = s->nb_taps - toffset;
const int size = remaining >= seg->part_size ? seg->part_size : remaining;
if (size < 8) {
for (n = 0; n < size; n++)
coeff[coffset + n].re = time[toffset + n];
toffset += size;
continue;
}
memset(blockin, 0, sizeof(*blockin) * seg->fft_length);
memcpy(blockin, time + toffset, size * sizeof(*blockin));
seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(float));
for (n = 0; n < seg->part_size + 1; n++) {
coeff[coffset + n].re = blockout[2 * n];
coeff[coffset + n].im = blockout[2 * n + 1];
}
toffset += size;
}
av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
}
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
convert_channels_float(ctx, s);
break;
case AV_SAMPLE_FMT_DBLP:
convert_channels_double(ctx, s);
break;
}
s->have_coeffs = 1;
@ -762,9 +437,10 @@ static int activate(AVFilterContext *ctx)
static int query_formats(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
static const enum AVSampleFormat sample_fmts[3][3] = {
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
};
static const enum AVPixelFormat pix_fmts[] = {
AV_PIX_FMT_RGB0,
@ -801,7 +477,7 @@ static int query_formats(AVFilterContext *ctx)
}
}
if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts)) < 0)
if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
@ -827,6 +503,7 @@ FF_ENABLE_DEPRECATION_WARNINGS
s->nb_channels = outlink->ch_layout.nb_channels;
s->nb_coef_channels = ctx->inputs[1 + s->selir]->ch_layout.nb_channels;
s->format = outlink->format;
return 0;
}
@ -977,6 +654,10 @@ static const AVOption afir_options[] = {
{ "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
{ "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
{ "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, "precision" },
{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
{ NULL }
};

99
libavfilter/af_afir.h Normal file
View File

@ -0,0 +1,99 @@
/*
* Copyright (c) 2017 Paul B Mahol
*
* 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
*/
#ifndef AVFILTER_AFIR_H
#define AVFILTER_AFIR_H
#include "libavutil/float_dsp.h"
#include "libavutil/frame.h"
#include "libavutil/rational.h"
#include "libavutil/tx.h"
#include "avfilter.h"
#include "af_afirdsp.h"
typedef struct AudioFIRSegment {
int nb_partitions;
int part_size;
int block_size;
int fft_length;
int coeff_size;
int input_size;
int input_offset;
int *output_offset;
int *part_index;
AVFrame *sumin;
AVFrame *sumout;
AVFrame *blockin;
AVFrame *blockout;
AVFrame *buffer;
AVFrame *coeff;
AVFrame *input;
AVFrame *output;
AVTXContext **tx, **itx;
av_tx_fn tx_fn, itx_fn;
} AudioFIRSegment;
typedef struct AudioFIRContext {
const AVClass *class;
float wet_gain;
float dry_gain;
float length;
int gtype;
float ir_gain;
int ir_format;
float max_ir_len;
int response;
int w, h;
AVRational frame_rate;
int ir_channel;
int minp;
int maxp;
int nb_irs;
int selir;
int precision;
int format;
double gain;
int eof_coeffs[32];
int have_coeffs;
int nb_taps;
int nb_channels;
int nb_coef_channels;
int one2many;
AudioFIRSegment seg[1024];
int nb_segments;
AVFrame *in;
AVFrame *ir[32];
AVFrame *video;
int min_part_size;
int64_t pts;
AudioFIRDSPContext afirdsp;
AVFloatDSPContext *fdsp;
} AudioFIRContext;
#endif /* AVFILTER_AFIR_H */

20
libavfilter/af_afirdsp.h
View File

@ -29,6 +29,8 @@
typedef struct AudioFIRDSPContext {
void (*fcmul_add)(float *sum, const float *t, const float *c,
ptrdiff_t len);
void (*dcmul_add)(double *sum, const double *t, const double *c,
ptrdiff_t len);
} AudioFIRDSPContext;
void ff_afir_init_x86(AudioFIRDSPContext *s);
@ -50,9 +52,27 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le
sum[2 * n] += t[2 * n] * c[2 * n];
}
static void dcmul_add_c(double *sum, const double *t, const double *c, ptrdiff_t len)
{
int n;
for (n = 0; n < len; n++) {
const double cre = c[2 * n ];
const double cim = c[2 * n + 1];
const double tre = t[2 * n ];
const double tim = t[2 * n + 1];
sum[2 * n ] += tre * cre - tim * cim;
sum[2 * n + 1] += tre * cim + tim * cre;
}
sum[2 * n] += t[2 * n] * c[2 * n];
}
static av_unused void ff_afir_init(AudioFIRDSPContext *dsp)
{
dsp->fcmul_add = fcmul_add_c;
dsp->dcmul_add = dcmul_add_c;
if (ARCH_X86)
ff_afir_init_x86(dsp);

392
libavfilter/afir_template.c Normal file
View File

@ -0,0 +1,392 @@
/*
* Copyright (c) 2017 Paul B Mahol
*
* 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
*/
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#include "audio.h"
#undef ctype
#undef ftype
#undef SQRT
#undef SAMPLE_FORMAT
#if DEPTH == 32
#define SAMPLE_FORMAT float
#define SQRT sqrtf
#define ctype AVComplexFloat
#define ftype float
#else
#define SAMPLE_FORMAT double
#define SQRT sqrt
#define ctype AVComplexDouble
#define ftype double
#endif
#define fn3(a,b) a##_##b
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
static void fn(draw_response)(AVFilterContext *ctx, AVFrame *out)
{
AudioFIRContext *s = ctx->priv;
ftype *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
ftype min_delay = FLT_MAX, max_delay = FLT_MIN;
int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
char text[32];
int channel, i, x;
memset(out->data[0], 0, s->h * out->linesize[0]);
phase = av_malloc_array(s->w, sizeof(*phase));
mag = av_malloc_array(s->w, sizeof(*mag));
delay = av_malloc_array(s->w, sizeof(*delay));
if (!mag || !phase || !delay)
goto end;
channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->ch_layout.nb_channels - 1);
for (i = 0; i < s->w; i++) {
const ftype *src = (const ftype *)s->ir[s->selir]->extended_data[channel];
double w = i * M_PI / (s->w - 1);
double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
for (x = 0; x < s->nb_taps; x++) {
real += cos(-x * w) * src[x];
imag += sin(-x * w) * src[x];
real_num += cos(-x * w) * src[x] * x;
imag_num += sin(-x * w) * src[x] * x;
}
mag[i] = hypot(real, imag);
phase[i] = atan2(imag, real);
div = real * real + imag * imag;
delay[i] = (real_num * real + imag_num * imag) / div;
min = fminf(min, mag[i]);
max = fmaxf(max, mag[i]);
min_delay = fminf(min_delay, delay[i]);
max_delay = fmaxf(max_delay, delay[i]);
}
for (i = 0; i < s->w; i++) {
int ymag = mag[i] / max * (s->h - 1);
int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
if (prev_ymag < 0)
prev_ymag = ymag;
if (prev_yphase < 0)
prev_yphase = yphase;
if (prev_ydelay < 0)
prev_ydelay = ydelay;
draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
prev_ymag = ymag;
prev_yphase = yphase;
prev_ydelay = ydelay;
}
if (s->w > 400 && s->h > 100) {
drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", max);
drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min);
drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", max_delay);
drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min_delay);
drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
}
end:
av_free(delay);
av_free(phase);
av_free(mag);
}
static void fn(convert_channels)(AVFilterContext *ctx, AudioFIRContext *s)
{
for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
int toffset = 0;
for (int i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
time[i] = 0;
av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
for (int segment = 0; segment < s->nb_segments; segment++) {
AudioFIRSegment *seg = &s->seg[segment];
ftype *blockin = (ftype *)seg->blockin->extended_data[ch];
ftype *blockout = (ftype *)seg->blockout->extended_data[ch];
ctype *coeff = (ctype *)seg->coeff->extended_data[ch];
av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
for (int i = 0; i < seg->nb_partitions; i++) {
const int coffset = i * seg->coeff_size;
const int remaining = s->nb_taps - toffset;
const int size = remaining >= seg->part_size ? seg->part_size : remaining;
if (size < 8) {
for (int n = 0; n < size; n++)
coeff[coffset + n].re = time[toffset + n];
toffset += size;
continue;
}
memset(blockin, 0, sizeof(*blockin) * seg->fft_length);
memcpy(blockin, time + toffset, size * sizeof(*blockin));
seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(ftype));
for (int n = 0; n < seg->part_size + 1; n++) {
coeff[coffset + n].re = blockout[2 * n];
coeff[coffset + n].im = blockout[2 * n + 1];
}
toffset += size;
}
av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
}
}
}
static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps)
{
ftype power = 0;
int ch;
switch (s->gtype) {
case -1:
/* nothing to do */
break;
case 0:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
for (int i = 0; i < cur_nb_taps; i++)
power += FFABS(time[i]);
}
s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
break;
case 1:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
for (int i = 0; i < cur_nb_taps; i++)
power += time[i];
}
s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
break;
case 2:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
for (int i = 0; i < cur_nb_taps; i++)
power += time[i] * time[i];
}
s->gain = SQRT(ch / power);
break;
default:
return AVERROR_BUG;
}
s->gain = FFMIN(s->gain * s->ir_gain, 1.);
av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
#if DEPTH == 32
s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
#else
s->fdsp->vector_dmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 8));
#endif
}
return 0;
}
static void fn(direct)(const ftype *in, const ctype *ir, int len, ftype *out)
{
for (int n = 0; n < len; n++)
for (int m = 0; m <= n; m++)
out[n] += ir[m].re * in[n - m];
}
static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples)
{
if ((nb_samples & 15) == 0 && nb_samples >= 16) {
#if DEPTH == 32
s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
#else
s->fdsp->vector_dmac_scalar(dst, src, 1.0, nb_samples);
#endif
} else {
for (int n = 0; n < nb_samples; n++)
dst[n] += src[n];
}
}
static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
{
AudioFIRContext *s = ctx->priv;
const ftype *in = (const ftype *)s->in->extended_data[ch] + offset;
ftype *blockin, *blockout, *buf, *ptr = (ftype *)out->extended_data[ch] + offset;
const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
int n, i, j;
for (int segment = 0; segment < s->nb_segments; segment++) {
AudioFIRSegment *seg = &s->seg[segment];
ftype *src = (ftype *)seg->input->extended_data[ch];
ftype *dst = (ftype *)seg->output->extended_data[ch];
ftype *sumin = (ftype *)seg->sumin->extended_data[ch];
ftype *sumout = (ftype *)seg->sumout->extended_data[ch];
if (s->min_part_size >= 8) {
#if DEPTH == 32
s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
#else
s->fdsp->vector_dmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 8));
#endif
emms_c();
} else {
for (n = 0; n < nb_samples; n++)
src[seg->input_offset + n] = in[n] * s->dry_gain;
}
seg->output_offset[ch] += s->min_part_size;
if (seg->output_offset[ch] == seg->part_size) {
seg->output_offset[ch] = 0;
} else {
memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
dst += seg->output_offset[ch];
fn(fir_fadd)(s, ptr, dst, nb_samples);
continue;
}
if (seg->part_size < 8) {
memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
j = seg->part_index[ch];
for (i = 0; i < seg->nb_partitions; i++) {
const int coffset = j * seg->coeff_size;
const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
fn(direct)(src, coeff, nb_samples, dst);
if (j == 0)
j = seg->nb_partitions;
j--;
}
seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
for (n = 0; n < nb_samples; n++) {
ptr[n] += dst[n];
}
continue;
}
memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
blockin = (ftype *)seg->blockin->extended_data[ch] + seg->part_index[ch] * seg->block_size;
blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
memset(blockin + seg->part_size, 0, sizeof(*blockin) * (seg->fft_length - seg->part_size));
memcpy(blockin, src, sizeof(*src) * seg->part_size);
seg->tx_fn(seg->tx[ch], blockout, blockin, sizeof(ftype));
j = seg->part_index[ch];
for (i = 0; i < seg->nb_partitions; i++) {
const int coffset = j * seg->coeff_size;
const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + i * seg->block_size;
const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
#if DEPTH == 32
s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size);
#else
s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size);
#endif
if (j == 0)
j = seg->nb_partitions;
j--;
}
seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ftype));
buf = (ftype *)seg->buffer->extended_data[ch];
fn(fir_fadd)(s, buf, sumout, seg->part_size);
memcpy(dst, buf, seg->part_size * sizeof(*dst));
buf = (ftype *)seg->buffer->extended_data[ch];
memcpy(buf, sumout + seg->part_size, seg->part_size * sizeof(*buf));
seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
fn(fir_fadd)(s, ptr, dst, nb_samples);
}
if (s->min_part_size >= 8) {
#if DEPTH == 32
s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
#else
s->fdsp->vector_dmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 8));
#endif
emms_c();
} else {
for (n = 0; n < nb_samples; n++)
ptr[n] *= s->wet_gain;
}
return 0;
}