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FFmpeg/libavfilter/asrc_sine.c
Marton Balint 8d6f3bcb96 avfilter/asrc_sine: increase frequency accuracy
Previously the delta phase was fixed point fractional with 2^32 fractions,
which caused inaccuracies in the output frequency, unless the input
frequency*2^32 was divisable by the sample rate.

This patch improves frequency accuracy by tracking subfractions of the delta
phase fractions. For this we are using a denominator which is a multiple of the
sample rate, making sure that integer frequencies are always accurately
represented.

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-11-29 21:16:22 +01:00

313 lines
10 KiB
C

/*
* Copyright (c) 2013 Nicolas George
*
* 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 <float.h>
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/eval.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
typedef struct SamplingContext {
uint32_t phi; ///< current phase of the sine (2pi = 1<<32)
uint32_t dphi; ///< phase increment between two samples
int phi_rem; ///< current fractional phase in 1/dphi_den subfractions
int dphi_rem;
int dphi_den;
} SamplingContext;
typedef struct SineContext {
const AVClass *class;
double frequency;
double beep_factor;
char *samples_per_frame;
AVExpr *samples_per_frame_expr;
int sample_rate;
int64_t duration;
int16_t *sin;
int64_t pts;
SamplingContext signal;
SamplingContext beep;
unsigned beep_period;
unsigned beep_index;
unsigned beep_length;
} SineContext;
#define CONTEXT SineContext
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
{ name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type, \
{ .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
#define OPT_INT(name, field, def, min, max, descr, ...) \
OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
#define OPT_DBL(name, field, def, min, max, descr, ...) \
OPT_GENERIC(name, field, def, min, max, descr, DOUBLE, dbl, __VA_ARGS__)
#define OPT_DUR(name, field, def, min, max, descr, ...) \
OPT_GENERIC(name, field, def, min, max, descr, DURATION, str, __VA_ARGS__)
#define OPT_STR(name, field, def, min, max, descr, ...) \
OPT_GENERIC(name, field, def, min, max, descr, STRING, str, __VA_ARGS__)
static const AVOption sine_options[] = {
OPT_DBL("frequency", frequency, 440, 0, DBL_MAX, "set the sine frequency",),
OPT_DBL("f", frequency, 440, 0, DBL_MAX, "set the sine frequency",),
OPT_DBL("beep_factor", beep_factor, 0, 0, DBL_MAX, "set the beep frequency factor",),
OPT_DBL("b", beep_factor, 0, 0, DBL_MAX, "set the beep frequency factor",),
OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
OPT_INT("r", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
OPT_DUR("duration", duration, 0, 0, INT64_MAX, "set the audio duration",),
OPT_DUR("d", duration, 0, 0, INT64_MAX, "set the audio duration",),
OPT_STR("samples_per_frame", samples_per_frame, "1024", 0, 0, "set the number of samples per frame",),
{NULL}
};
AVFILTER_DEFINE_CLASS(sine);
#define LOG_PERIOD 15
#define AMPLITUDE 4095
#define AMPLITUDE_SHIFT 3
static void make_sin_table(int16_t *sin)
{
unsigned half_pi = 1 << (LOG_PERIOD - 2);
unsigned ampls = AMPLITUDE << AMPLITUDE_SHIFT;
uint64_t unit2 = (uint64_t)(ampls * ampls) << 32;
unsigned step, i, c, s, k, new_k, n2;
/* Principle: if u = exp(i*a1) and v = exp(i*a2), then
exp(i*(a1+a2)/2) = (u+v) / length(u+v) */
sin[0] = 0;
sin[half_pi] = ampls;
for (step = half_pi; step > 1; step /= 2) {
/* k = (1 << 16) * amplitude / length(u+v)
In exact values, k is constant at a given step */
k = 0x10000;
for (i = 0; i < half_pi / 2; i += step) {
s = sin[i] + sin[i + step];
c = sin[half_pi - i] + sin[half_pi - i - step];
n2 = s * s + c * c;
/* Newton's method to solve n² * k² = unit² */
while (1) {
new_k = (k + unit2 / ((uint64_t)k * n2) + 1) >> 1;
if (k == new_k)
break;
k = new_k;
}
sin[i + step / 2] = (k * s + 0x7FFF) >> 16;
sin[half_pi - i - step / 2] = (k * c + 0x8000) >> 16;
}
}
/* Unshift amplitude */
for (i = 0; i <= half_pi; i++)
sin[i] = (sin[i] + (1 << (AMPLITUDE_SHIFT - 1))) >> AMPLITUDE_SHIFT;
/* Use symmetries to fill the other three quarters */
for (i = 0; i < half_pi; i++)
sin[half_pi * 2 - i] = sin[i];
for (i = 0; i < 2 * half_pi; i++)
sin[i + 2 * half_pi] = -sin[i];
}
static const char *const var_names[] = {
"n",
"pts",
"t",
"TB",
NULL
};
enum {
VAR_N,
VAR_PTS,
VAR_T,
VAR_TB,
VAR_VARS_NB
};
static void sampling_init(SamplingContext *c, double frequency, int sample_rate)
{
AVRational r;
int r_den, max_r_den;
max_r_den = INT_MAX / sample_rate;
frequency = fmod(frequency, sample_rate);
r = av_d2q(fmod(frequency, 1.0), max_r_den);
r_den = FFMIN(r.den, max_r_den);
c->dphi = ldexp(frequency, 32) / sample_rate;
c->dphi_den = r_den * sample_rate;
c->dphi_rem = round((ldexp(frequency, 32) / sample_rate - c->dphi) * c->dphi_den);
if (c->dphi_rem >= c->dphi_den) {
c->dphi++;
c->dphi_rem = 0;
}
c->phi_rem = (-c->dphi_den - 1) / 2;
}
static av_always_inline void sampling_advance(SamplingContext *c)
{
c->phi += c->dphi;
c->phi_rem += c->dphi_rem;
if (c->phi_rem >= 0) {
c->phi_rem -= c->dphi_den;
c->phi++;
}
}
static av_cold int init(AVFilterContext *ctx)
{
int ret;
SineContext *sine = ctx->priv;
if (!(sine->sin = av_malloc(sizeof(*sine->sin) << LOG_PERIOD)))
return AVERROR(ENOMEM);
sampling_init(&sine->signal, sine->frequency, sine->sample_rate);
make_sin_table(sine->sin);
if (sine->beep_factor) {
sine->beep_period = sine->sample_rate;
sine->beep_length = sine->beep_period / 25;
sampling_init(&sine->beep, sine->beep_factor * sine->frequency, sine->sample_rate);
}
ret = av_expr_parse(&sine->samples_per_frame_expr,
sine->samples_per_frame, var_names,
NULL, NULL, NULL, NULL, 0, sine);
if (ret < 0)
return ret;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
SineContext *sine = ctx->priv;
av_expr_free(sine->samples_per_frame_expr);
sine->samples_per_frame_expr = NULL;
av_freep(&sine->sin);
}
static av_cold int query_formats(const AVFilterContext *ctx,
AVFilterFormatsConfig **cfg_in,
AVFilterFormatsConfig **cfg_out)
{
const SineContext *sine = ctx->priv;
static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
int sample_rates[] = { sine->sample_rate, -1 };
static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE };
int ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, chlayouts);
if (ret < 0)
return ret;
return ff_set_common_samplerates_from_list2(ctx, cfg_in, cfg_out, sample_rates);
}
static av_cold int config_props(AVFilterLink *outlink)
{
SineContext *sine = outlink->src->priv;
sine->duration = av_rescale(sine->duration, sine->sample_rate, AV_TIME_BASE);
return 0;
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *outlink = ctx->outputs[0];
FilterLink *outl = ff_filter_link(outlink);
SineContext *sine = ctx->priv;
AVFrame *frame;
double values[VAR_VARS_NB] = {
[VAR_N] = outl->frame_count_in,
[VAR_PTS] = sine->pts,
[VAR_T] = sine->pts * av_q2d(outlink->time_base),
[VAR_TB] = av_q2d(outlink->time_base),
};
int i, nb_samples = lrint(av_expr_eval(sine->samples_per_frame_expr, values, sine));
int16_t *samples;
if (!ff_outlink_frame_wanted(outlink))
return FFERROR_NOT_READY;
if (nb_samples <= 0) {
av_log(sine, AV_LOG_WARNING, "nb samples expression evaluated to %d, "
"defaulting to 1024\n", nb_samples);
nb_samples = 1024;
}
if (sine->duration) {
nb_samples = FFMIN(nb_samples, sine->duration - sine->pts);
av_assert1(nb_samples >= 0);
if (!nb_samples) {
ff_outlink_set_status(outlink, AVERROR_EOF, sine->pts);
return 0;
}
}
if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
return AVERROR(ENOMEM);
samples = (int16_t *)frame->data[0];
for (i = 0; i < nb_samples; i++) {
samples[i] = sine->sin[sine->signal.phi >> (32 - LOG_PERIOD)];
sampling_advance(&sine->signal);
if (sine->beep_index < sine->beep_length) {
samples[i] += sine->sin[sine->beep.phi >> (32 - LOG_PERIOD)] * 2;
sampling_advance(&sine->beep);
}
if (++sine->beep_index == sine->beep_period)
sine->beep_index = 0;
}
frame->pts = sine->pts;
sine->pts += nb_samples;
return ff_filter_frame(outlink, frame);
}
static const AVFilterPad sine_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
},
};
const AVFilter ff_asrc_sine = {
.name = "sine",
.description = NULL_IF_CONFIG_SMALL("Generate sine wave audio signal."),
.init = init,
.uninit = uninit,
.activate = activate,
.priv_size = sizeof(SineContext),
.inputs = NULL,
FILTER_OUTPUTS(sine_outputs),
FILTER_QUERY_FUNC2(query_formats),
.priv_class = &sine_class,
};