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FFmpeg/libavfilter/af_afir.c
Paul B Mahol c337b0f826 avfilter/af_afir: remove pts rewriting code 
This is not needed at all.
2022-02-22 13:25:21 +01:00

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/*
* 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
*/
/**
* @file
* An arbitrary audio FIR filter
*/
#include <float.h>
#include "libavutil/tx.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/opt.h"
#include "libavutil/xga_font_data.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "internal.h"
#include "af_afir.h"
static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
{
int n;
for (n = 0; n < len; n++) {
const float cre = c[2 * n ];
const float cim = c[2 * n + 1];
const float tre = t[2 * n ];
const float 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 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->channels * jobnr) / nb_jobs;
const int end = (out->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->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;
int font_height;
int i;
font = avpriv_cga_font, font_height = 8;
for (i = 0; txt[i]; i++) {
int char_y, mask;
uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
for (char_y = 0; char_y < font_height; char_y++) {
for (mask = 0x80; mask; mask >>= 1) {
if (font[txt[i] * font_height + char_y] & mask)
AV_WL32(p, color);
p += 4;
}
p += pic->linesize[0] - 8 * 4;
}
}
}
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
{
int dx = FFABS(x1-x0);
int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
int err = (dx>dy ? dx : -dy) / 2, e2;
for (;;) {
AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
if (x0 == x1 && y0 == y1)
break;
e2 = err;
if (e2 >-dx) {
err -= dy;
x0--;
}
if (e2 < dy) {
err += dx;
y0 += sy;
}
}
}
static void draw_response(AVFilterContext *ctx, AVFrame *out)
{
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]->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;
}
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 int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
int offset, int nb_partitions, int part_size)
{
AudioFIRContext *s = ctx->priv;
seg->tx = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->tx));
seg->itx = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->itx));
if (!seg->tx || !seg->itx)
return AVERROR(ENOMEM);
seg->fft_length = part_size * 2 + 1;
seg->part_size = part_size;
seg->block_size = FFALIGN(seg->fft_length, 32);
seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
seg->nb_partitions = nb_partitions;
seg->input_size = offset + s->min_part_size;
seg->input_offset = offset;
seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
if (!seg->part_index || !seg->output_offset)
return AVERROR(ENOMEM);
for (int ch = 0; ch < ctx->inputs[0]->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);
if (!seg->tx[ch] || !seg->itx[ch])
return AVERROR(ENOMEM);
}
seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
seg->blockin = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
seg->coeff = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);
seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockin || !seg->blockout || !seg->coeff || !seg->input || !seg->output)
return AVERROR(ENOMEM);
return 0;
}
static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
{
AudioFIRContext *s = ctx->priv;
if (seg->tx) {
for (int ch = 0; ch < s->nb_channels; ch++) {
av_tx_uninit(&seg->tx[ch]);
}
}
av_freep(&seg->tx);
if (seg->itx) {
for (int ch = 0; ch < s->nb_channels; ch++) {
av_tx_uninit(&seg->itx[ch]);
}
}
av_freep(&seg->itx);
av_freep(&seg->output_offset);
av_freep(&seg->part_index);
av_frame_free(&seg->blockin);
av_frame_free(&seg->blockout);
av_frame_free(&seg->sumin);
av_frame_free(&seg->sumout);
av_frame_free(&seg->buffer);
av_frame_free(&seg->coeff);
av_frame_free(&seg->input);
av_frame_free(&seg->output);
seg->input_size = 0;
}
static int convert_coeffs(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
int ret, i, ch, n, cur_nb_taps;
float power = 0;
if (!s->nb_taps) {
int part_size, max_part_size;
int left, offset = 0;
s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
if (s->nb_taps <= 0)
return AVERROR(EINVAL);
if (s->minp > s->maxp) {
s->maxp = s->minp;
}
left = s->nb_taps;
part_size = 1 << av_log2(s->minp);
max_part_size = 1 << av_log2(s->maxp);
s->min_part_size = part_size;
for (i = 0; left > 0; i++) {
int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
s->nb_segments = i + 1;
ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
if (ret < 0)
return ret;
offset += nb_partitions * part_size;
left -= nb_partitions * part_size;
part_size *= 2;
part_size = FFMIN(part_size, max_part_size);
}
}
if (!s->ir[s->selir]) {
ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);
if (ret < 0)
return ret;
if (ret == 0)
return AVERROR_BUG;
}
if (s->response)
draw_response(ctx, s->video);
s->gain = 1;
cur_nb_taps = s->ir[s->selir]->nb_samples;
switch (s->gtype) {
case -1:
/* nothing to do */
break;
case 0:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->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]->channels / power;
break;
case 1:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->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]->channels / power;
break;
case 2:
for (ch = 0; ch < ctx->inputs[1 + s->selir]->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]->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));
}
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]->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);
}
}
s->have_coeffs = 1;
return 0;
}
static int check_ir(AVFilterLink *link)
{
AVFilterContext *ctx = link->dst;
AudioFIRContext *s = ctx->priv;
int nb_taps, max_nb_taps;
nb_taps = ff_inlink_queued_samples(link);
max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
if (nb_taps > max_nb_taps) {
av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
return AVERROR(EINVAL);
}
return 0;
}
static int activate(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret, status, available, wanted;
AVFrame *in = NULL;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
if (s->response)
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
if (!s->eof_coeffs[s->selir]) {
ret = check_ir(ctx->inputs[1 + s->selir]);
if (ret < 0)
return ret;
if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
s->eof_coeffs[s->selir] = 1;
if (!s->eof_coeffs[s->selir]) {
if (ff_outlink_frame_wanted(ctx->outputs[0]))
ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
return 0;
}
}
if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
ret = convert_coeffs(ctx);
if (ret < 0)
return ret;
}
available = ff_inlink_queued_samples(ctx->inputs[0]);
wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
if (ret > 0)
ret = fir_frame(s, in, outlink);
if (ret < 0)
return ret;
if (s->response && s->have_coeffs) {
int64_t old_pts = s->video->pts;
int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
AVFrame *clone;
s->video->pts = new_pts;
clone = av_frame_clone(s->video);
if (!clone)
return AVERROR(ENOMEM);
return ff_filter_frame(ctx->outputs[1], clone);
}
}
if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
ff_filter_set_ready(ctx, 10);
return 0;
}
if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
if (status == AVERROR_EOF) {
ff_outlink_set_status(ctx->outputs[0], status, pts);
if (s->response)
ff_outlink_set_status(ctx->outputs[1], status, pts);
return 0;
}
}
if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
!ff_outlink_get_status(ctx->inputs[0])) {
ff_inlink_request_frame(ctx->inputs[0]);
return 0;
}
if (s->response &&
ff_outlink_frame_wanted(ctx->outputs[1]) &&
!ff_outlink_get_status(ctx->inputs[0])) {
ff_inlink_request_frame(ctx->inputs[0]);
return 0;
}
return FFERROR_NOT_READY;
}
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 AVPixelFormat pix_fmts[] = {
AV_PIX_FMT_RGB0,
AV_PIX_FMT_NONE
};
int ret;
if (s->response) {
AVFilterLink *videolink = ctx->outputs[1];
AVFilterFormats *formats = ff_make_format_list(pix_fmts);
if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
return ret;
}
if (s->ir_format) {
ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
} else {
AVFilterChannelLayouts *mono = NULL;
AVFilterChannelLayouts *layouts = ff_all_channel_counts();
if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
return ret;
if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
return ret;
ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
if (ret)
return ret;
for (int i = 1; i < ctx->nb_inputs; i++) {
if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
return ret;
}
}
if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts)) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFIRContext *s = ctx->priv;
s->one2many = ctx->inputs[1 + s->selir]->channels == 1;
outlink->sample_rate = ctx->inputs[0]->sample_rate;
outlink->time_base = ctx->inputs[0]->time_base;
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
s->nb_channels = outlink->channels;
s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
for (int i = 0; i < s->nb_segments; i++) {
uninit_segment(ctx, &s->seg[i]);
}
av_freep(&s->fdsp);
for (int i = 0; i < s->nb_irs; i++) {
av_frame_free(&s->ir[i]);
}
av_frame_free(&s->video);
}
static int config_video(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFIRContext *s = ctx->priv;
outlink->sample_aspect_ratio = (AVRational){1,1};
outlink->w = s->w;
outlink->h = s->h;
outlink->frame_rate = s->frame_rate;
outlink->time_base = av_inv_q(outlink->frame_rate);
av_frame_free(&s->video);
s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
if (!s->video)
return AVERROR(ENOMEM);
return 0;
}
void ff_afir_init(AudioFIRDSPContext *dsp)
{
dsp->fcmul_add = fcmul_add_c;
if (ARCH_X86)
ff_afir_init_x86(dsp);
}
static av_cold int init(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
AVFilterPad pad, vpad;
int ret;
pad = (AVFilterPad) {
.name = "main",
.type = AVMEDIA_TYPE_AUDIO,
};
ret = ff_append_inpad(ctx, &pad);
if (ret < 0)
return ret;
for (int n = 0; n < s->nb_irs; n++) {
pad = (AVFilterPad) {
.name = av_asprintf("ir%d", n),
.type = AVMEDIA_TYPE_AUDIO,
};
if (!pad.name)
return AVERROR(ENOMEM);
ret = ff_append_inpad_free_name(ctx, &pad);
if (ret < 0)
return ret;
}
pad = (AVFilterPad) {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
};
ret = ff_append_outpad(ctx, &pad);
if (ret < 0)
return ret;
if (s->response) {
vpad = (AVFilterPad){
.name = "filter_response",
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video,
};
ret = ff_append_outpad(ctx, &vpad);
if (ret < 0)
return ret;
}
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
ff_afir_init(&s->afirdsp);
return 0;
}
static int process_command(AVFilterContext *ctx,
const char *cmd,
const char *arg,
char *res,
int res_len,
int flags)
{
AudioFIRContext *s = ctx->priv;
int prev_ir = s->selir;
int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
if (ret < 0)
return ret;
s->selir = FFMIN(s->nb_irs - 1, s->selir);
if (prev_ir != s->selir) {
s->have_coeffs = 0;
}
return 0;
}
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
#define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define OFFSET(x) offsetof(AudioFIRContext, x)
static const AVOption afir_options[] = {
{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
{ "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
{ "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
{ "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
{ "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
{ "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
{ "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
{ "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
{ "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
{ "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
{ "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
{ "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
{ "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
{ "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
{ "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
{ "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 32768, AF },
{ "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 },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afir);
const AVFilter ff_af_afir = {
.name = "afir",
.description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
.priv_size = sizeof(AudioFIRContext),
.priv_class = &afir_class,
FILTER_QUERY_FUNC(query_formats),
.init = init,
.activate = activate,
.uninit = uninit,
.process_command = process_command,
.flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SLICE_THREADS,
};
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