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FFmpeg/libavfilter/asrc_afirsrc.c

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/*
* Copyright (c) 2020 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 "libavutil/channel_layout.h"
#include "libavutil/eval.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
#include "window_func.h"
typedef struct AudioFIRSourceContext {
const AVClass *class;
char *freq_points_str;
char *magnitude_str;
char *phase_str;
int nb_taps;
int sample_rate;
int nb_samples;
int win_func;
AVComplexFloat *complexf;
float *freq;
float *magnitude;
float *phase;
int freq_size;
int magnitude_size;
int phase_size;
int nb_freq;
int nb_magnitude;
int nb_phase;
float *taps;
float *win;
int64_t pts;
AVTXContext *tx_ctx;
av_tx_fn tx_fn;
} AudioFIRSourceContext;
#define OFFSET(x) offsetof(AudioFIRSourceContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption afirsrc_options[] = {
{ "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
{ "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
{ "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
{ "f", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
{ "magnitude", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
{ "m", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
{ "phase", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
{ "p", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
{ "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
{NULL}
};
AVFILTER_DEFINE_CLASS(afirsrc);
static av_cold int init(AVFilterContext *ctx)
{
AudioFIRSourceContext *s = ctx->priv;
if (!(s->nb_taps & 1)) {
av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps);
s->nb_taps |= 1;
}
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioFIRSourceContext *s = ctx->priv;
av_freep(&s->win);
av_freep(&s->taps);
av_freep(&s->freq);
av_freep(&s->magnitude);
av_freep(&s->phase);
av_freep(&s->complexf);
av_tx_uninit(&s->tx_ctx);
}
static av_cold int query_formats(AVFilterContext *ctx)
{
AudioFIRSourceContext *s = ctx->priv;
static const int64_t chlayouts[] = { AV_CH_LAYOUT_MONO, -1 };
int sample_rates[] = { s->sample_rate, -1 };
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
if (ret < 0)
return ret;
return ff_set_common_samplerates_from_list(ctx, sample_rates);
}
static int parse_string(char *str, float **items, int *nb_items, int *items_size)
{
float *new_items;
char *tail;
new_items = av_fast_realloc(NULL, items_size, 1 * sizeof(float));
if (!new_items)
return AVERROR(ENOMEM);
*items = new_items;
tail = str;
if (!tail)
return AVERROR(EINVAL);
do {
(*items)[(*nb_items)++] = av_strtod(tail, &tail);
new_items = av_fast_realloc(*items, items_size, (*nb_items + 1) * sizeof(float));
if (!new_items)
return AVERROR(ENOMEM);
*items = new_items;
if (tail && *tail)
tail++;
} while (tail && *tail);
return 0;
}
static void lininterp(AVComplexFloat *complexf,
const float *freq,
const float *magnitude,
const float *phase,
int m, int minterp)
{
for (int i = 0; i < minterp; i++) {
for (int j = 1; j < m; j++) {
const float x = i / (float)minterp;
if (x <= freq[j]) {
const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];
const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];
complexf[i].re = mg * cosf(ph);
complexf[i].im = mg * sinf(ph);
break;
}
}
}
}
static av_cold int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFIRSourceContext *s = ctx->priv;
float overlap, scale = 1.f, compensation;
int fft_size, middle, ret;
s->nb_freq = s->nb_magnitude = s->nb_phase = 0;
ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
if (ret < 0)
return ret;
ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
if (ret < 0)
return ret;
ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size);
if (ret < 0)
return ret;
if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) {
av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n");
return AVERROR(EINVAL);
}
for (int i = 0; i < s->nb_freq; i++) {
if (i == 0 && s->freq[i] != 0.f) {
av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n");
return AVERROR(EINVAL);
}
if (i == s->nb_freq - 1 && s->freq[i] != 1.f) {
av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n");
return AVERROR(EINVAL);
}
if (i && s->freq[i] < s->freq[i-1]) {
av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n");
return AVERROR(EINVAL);
}
}
fft_size = 1 << (av_log2(s->nb_taps) + 1);
s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf));
if (!s->complexf)
return AVERROR(ENOMEM);
ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
if (ret < 0)
return ret;
s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
if (!s->taps)
return AVERROR(ENOMEM);
s->win = av_calloc(s->nb_taps, sizeof(*s->win));
if (!s->win)
return AVERROR(ENOMEM);
generate_window_func(s->win, s->nb_taps, s->win_func, &overlap);
lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2);
s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(float));
compensation = 2.f / fft_size;
middle = s->nb_taps / 2;
for (int i = 0; i <= middle; i++) {
s->taps[ i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i];
s->taps[middle + i] = s->complexf[fft_size + i].re * compensation * s->win[middle + i];
}
s->pts = 0;
return 0;
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *outlink = ctx->outputs[0];
AudioFIRSourceContext *s = ctx->priv;
AVFrame *frame;
int nb_samples;
if (!ff_outlink_frame_wanted(outlink))
return FFERROR_NOT_READY;
nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
if (nb_samples <= 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}
if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
return AVERROR(ENOMEM);
memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
frame->pts = s->pts;
s->pts += nb_samples;
return ff_filter_frame(outlink, frame);
}
static const AVFilterPad afirsrc_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_asrc_afirsrc = {
.name = "afirsrc",
.description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."),
.init = init,
.uninit = uninit,
.activate = activate,
.priv_size = sizeof(AudioFIRSourceContext),
.inputs = NULL,
FILTER_OUTPUTS(afirsrc_outputs),
FILTER_QUERY_FUNC(query_formats),
.priv_class = &afirsrc_class,
};
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