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FFmpeg/libavfilter/af_afir.c
Paul B Mahol 876101cf41 avfilter/af_afir: switch to activate
2018-09-15 21:59:01 +02:00

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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/audio_fifo.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 "libavcodec/avfft.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 int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
AudioFIRContext *s = ctx->priv;
const float *src = (const float *)s->in[0]->extended_data[ch];
int index1 = (s->index + 1) % 3;
int index2 = (s->index + 2) % 3;
float *sum = s->sum[ch];
AVFrame *out = arg;
float *block;
float *dst;
int n, i, j;
memset(sum, 0, sizeof(*sum) * s->fft_length);
block = s->block[ch] + s->part_index * s->block_size;
memset(block, 0, sizeof(*block) * s->fft_length);
s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, FFALIGN(s->nb_samples, 4));
emms_c();
av_rdft_calc(s->rdft[ch], block);
block[2 * s->part_size] = block[1];
block[1] = 0;
j = s->part_index;
for (i = 0; i < s->nb_partitions; i++) {
const int coffset = i * s->coeff_size;
const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
block = s->block[ch] + j * s->block_size;
s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
if (j == 0)
j = s->nb_partitions;
j--;
}
sum[1] = sum[2 * s->part_size];
av_rdft_calc(s->irdft[ch], sum);
dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
for (n = 0; n < s->part_size; n++) {
dst[n] += sum[n];
}
dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
if (out) {
float *ptr = (float *)out->extended_data[ch];
s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
emms_c();
}
return 0;
}
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFrame *out = NULL;
int ret;
s->nb_samples = in->nb_samples;
if (!s->want_skip) {
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
}
if (s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;
s->in[0] = in;
ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
s->part_index = (s->part_index + 1) % s->nb_partitions;
if (!s->want_skip) {
out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
}
s->index++;
if (s->index == 3)
s->index = 0;
av_frame_free(&in);
if (s->want_skip == 1) {
s->want_skip = 0;
ret = 0;
} else {
ret = ff_filter_frame(outlink, out);
}
return ret;
}
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, min = FLT_MAX, max = FLT_MIN;
int prev_ymag = -1, prev_yphase = -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));
if (!mag || !phase)
goto end;
channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
for (i = 0; i < s->w; i++) {
const float *src = (const float *)s->in[1]->extended_data[channel];
double w = i * M_PI / (s->w - 1);
double real = 0.;
double imag = 0.;
for (x = 0; x < s->nb_taps; x++) {
real += cos(-x * w) * src[x];
imag += sin(-x * w) * src[x];
}
mag[i] = hypot(real, imag);
phase[i] = atan2(imag, real);
min = fminf(min, mag[i]);
max = fmaxf(max, mag[i]);
}
for (i = 0; i < s->w; i++) {
int ymag = mag[i] / max * (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);
if (prev_ymag < 0)
prev_ymag = ymag;
if (prev_yphase < 0)
prev_yphase = yphase;
draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
prev_ymag = ymag;
prev_yphase = yphase;
}
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);
}
end:
av_free(phase);
av_free(mag);
}
static int convert_coeffs(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
int i, ch, n, N;
s->nb_taps = av_audio_fifo_size(s->fifo);
if (s->nb_taps <= 0)
return AVERROR(EINVAL);
for (n = 4; (1 << n) < s->nb_taps; n++);
N = FFMIN(n, 16);
s->ir_length = 1 << n;
s->fft_length = (1 << (N + 1)) + 1;
s->part_size = 1 << (N - 1);
s->block_size = FFALIGN(s->fft_length, 32);
s->coeff_size = FFALIGN(s->part_size + 1, 32);
s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
s->nb_coeffs = s->ir_length + s->nb_partitions;
for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
if (!s->sum[ch])
return AVERROR(ENOMEM);
}
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
if (!s->coeff[ch])
return AVERROR(ENOMEM);
}
for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
if (!s->block[ch])
return AVERROR(ENOMEM);
}
for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
s->rdft[ch] = av_rdft_init(N, DFT_R2C);
s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
if (!s->rdft[ch] || !s->irdft[ch])
return AVERROR(ENOMEM);
}
s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
if (!s->in[1])
return AVERROR(ENOMEM);
s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
if (!s->buffer)
return AVERROR(ENOMEM);
av_audio_fifo_read(s->fifo, (void **)s->in[1]->extended_data, s->nb_taps);
if (s->response)
draw_response(ctx, s->video);
if (s->again) {
float power = 0;
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
for (i = 0; i < s->nb_taps; i++)
power += FFABS(time[i]);
}
s->gain = sqrtf(1.f / (ctx->inputs[1]->channels * power)) / (sqrtf(ctx->inputs[1]->channels));
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
}
}
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
float *block = s->block[ch];
FFTComplex *coeff = s->coeff[ch];
for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
time[i] = 0;
for (i = 0; i < s->nb_partitions; i++) {
const float scale = 1.f / s->part_size;
const int toffset = i * s->part_size;
const int coffset = i * s->coeff_size;
const int boffset = s->part_size;
const int remaining = s->nb_taps - (i * s->part_size);
const int size = remaining >= s->part_size ? s->part_size : remaining;
memset(block, 0, sizeof(*block) * s->fft_length);
memcpy(block + boffset, time + toffset, size * sizeof(*block));
av_rdft_calc(s->rdft[0], block);
coeff[coffset].re = block[0] * scale;
coeff[coffset].im = 0;
for (n = 1; n < s->part_size; n++) {
coeff[coffset + n].re = block[2 * n] * scale;
coeff[coffset + n].im = block[2 * n + 1] * scale;
}
coeff[coffset + s->part_size].re = block[1] * scale;
coeff[coffset + s->part_size].im = 0;
}
}
av_frame_free(&s->in[1]);
av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
s->have_coeffs = 1;
return 0;
}
static int read_ir(AVFilterLink *link, AVFrame *frame)
{
AVFilterContext *ctx = link->dst;
AudioFIRContext *s = ctx->priv;
int nb_taps, max_nb_taps, ret;
ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data,
frame->nb_samples);
av_frame_free(&frame);
if (ret < 0)
return ret;
nb_taps = av_audio_fifo_size(s->fifo);
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];
AVFrame *in = NULL;
int ret, status;
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) {
AVFrame *ir = NULL;
if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &ir)) > 0) {
ret = read_ir(ctx->inputs[1], ir);
if (ret < 0)
return ret;
}
if (ret < 0)
return ret;
if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
if (status == AVERROR_EOF) {
s->eof_coeffs = 1;
}
}
if (!s->eof_coeffs) {
if (ff_outlink_frame_wanted(ctx->outputs[0]))
ff_inlink_request_frame(ctx->inputs[1]);
return 0;
}
}
if (!s->have_coeffs && s->eof_coeffs) {
ret = convert_coeffs(ctx);
if (ret < 0)
return ret;
}
if (s->need_padding) {
in = ff_get_audio_buffer(outlink, s->part_size);
if (!in)
return AVERROR(ENOMEM);
s->need_padding = 0;
ret = 1;
} else {
ret = ff_inlink_consume_samples(ctx->inputs[0], s->part_size, s->part_size, &in);
}
if (ret > 0) {
ret = fir_frame(s, in, outlink);
if (ret < 0)
return ret;
}
if (ret < 0)
return ret;
if (s->response && s->have_coeffs) {
if (ff_outlink_frame_wanted(ctx->outputs[1])) {
s->video->pts = s->pts;
ret = ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
if (ret < 0)
return ret;
}
}
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_inlink_request_frame(ctx->inputs[0]);
return 0;
}
if (s->response && ff_outlink_frame_wanted(ctx->outputs[1])) {
ff_inlink_request_frame(ctx->inputs[0]);
return 0;
}
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
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, i;
if (s->response) {
AVFilterLink *videolink = ctx->outputs[1];
formats = ff_make_format_list(pix_fmts);
if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
return ret;
}
layouts = ff_all_channel_counts();
if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
return ret;
for (i = 0; i < 2; i++) {
layouts = ff_all_channel_counts();
if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
return ret;
}
formats = ff_make_format_list(sample_fmts);
if ((ret = ff_set_common_formats(ctx, formats)) < 0)
return ret;
formats = ff_all_samplerates();
return ff_set_common_samplerates(ctx, formats);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFIRContext *s = ctx->priv;
if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
ctx->inputs[1]->channels != 1) {
av_log(ctx, AV_LOG_ERROR,
"Second input must have same number of channels as first input or "
"exactly 1 channel.\n");
return AVERROR(EINVAL);
}
s->one2many = ctx->inputs[1]->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->fifo = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
if (!s->fifo)
return AVERROR(ENOMEM);
s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
return AVERROR(ENOMEM);
s->nb_channels = outlink->channels;
s->nb_coef_channels = ctx->inputs[1]->channels;
s->want_skip = 1;
s->need_padding = 1;
s->pts = AV_NOPTS_VALUE;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
int ch;
if (s->sum) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_freep(&s->sum[ch]);
}
}
av_freep(&s->sum);
if (s->coeff) {
for (ch = 0; ch < s->nb_coef_channels; ch++) {
av_freep(&s->coeff[ch]);
}
}
av_freep(&s->coeff);
if (s->block) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_freep(&s->block[ch]);
}
}
av_freep(&s->block);
if (s->rdft) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_rdft_end(s->rdft[ch]);
}
}
av_freep(&s->rdft);
if (s->irdft) {
for (ch = 0; ch < s->nb_channels; ch++) {
av_rdft_end(s->irdft[ch]);
}
}
av_freep(&s->irdft);
av_frame_free(&s->in[1]);
av_frame_free(&s->buffer);
av_audio_fifo_free(s->fifo);
av_freep(&s->fdsp);
for (int i = 0; i < ctx->nb_outputs; i++)
av_freep(&ctx->output_pads[i].name);
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;
av_frame_free(&s->video);
s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
if (!s->video)
return AVERROR(ENOMEM);
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
AVFilterPad pad, vpad;
int ret;
pad = (AVFilterPad){
.name = av_strdup("default"),
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
};
if (!pad.name)
return AVERROR(ENOMEM);
if (s->response) {
vpad = (AVFilterPad){
.name = av_strdup("filter_response"),
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video,
};
if (!vpad.name)
return AVERROR(ENOMEM);
}
ret = ff_insert_outpad(ctx, 0, &pad);
if (ret < 0) {
av_freep(&pad.name);
return ret;
}
if (s->response) {
ret = ff_insert_outpad(ctx, 1, &vpad);
if (ret < 0) {
av_freep(&vpad.name);
return ret;
}
}
s->fcmul_add = fcmul_add_c;
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
if (ARCH_X86)
ff_afir_init_x86(s);
return 0;
}
static const AVFilterPad afir_inputs[] = {
{
.name = "main",
.type = AVMEDIA_TYPE_AUDIO,
},{
.name = "ir",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_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 },
{ "again", "enable auto gain", OFFSET(again), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
{ "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 },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afir);
AVFilter ff_af_afir = {
.name = "afir",
.description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
.priv_size = sizeof(AudioFIRContext),
.priv_class = &afir_class,
.query_formats = query_formats,
.init = init,
.activate = activate,
.uninit = uninit,
.inputs = afir_inputs,
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SLICE_THREADS,
};
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