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https://github.com/FFmpeg/FFmpeg.git
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8d6f3bcb96
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>
313 lines
10 KiB
C
313 lines
10 KiB
C
/*
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* Copyright (c) 2013 Nicolas George
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public License
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* as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public License
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* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <float.h>
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "formats.h"
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typedef struct SamplingContext {
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uint32_t phi; ///< current phase of the sine (2pi = 1<<32)
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uint32_t dphi; ///< phase increment between two samples
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int phi_rem; ///< current fractional phase in 1/dphi_den subfractions
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int dphi_rem;
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int dphi_den;
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} SamplingContext;
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typedef struct SineContext {
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const AVClass *class;
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double frequency;
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double beep_factor;
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char *samples_per_frame;
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AVExpr *samples_per_frame_expr;
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int sample_rate;
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int64_t duration;
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int16_t *sin;
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int64_t pts;
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SamplingContext signal;
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SamplingContext beep;
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unsigned beep_period;
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unsigned beep_index;
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unsigned beep_length;
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} SineContext;
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#define CONTEXT SineContext
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
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{ name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type, \
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{ .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
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#define OPT_INT(name, field, def, min, max, descr, ...) \
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OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
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#define OPT_DBL(name, field, def, min, max, descr, ...) \
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OPT_GENERIC(name, field, def, min, max, descr, DOUBLE, dbl, __VA_ARGS__)
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#define OPT_DUR(name, field, def, min, max, descr, ...) \
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OPT_GENERIC(name, field, def, min, max, descr, DURATION, str, __VA_ARGS__)
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#define OPT_STR(name, field, def, min, max, descr, ...) \
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OPT_GENERIC(name, field, def, min, max, descr, STRING, str, __VA_ARGS__)
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static const AVOption sine_options[] = {
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OPT_DBL("frequency", frequency, 440, 0, DBL_MAX, "set the sine frequency",),
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OPT_DBL("f", frequency, 440, 0, DBL_MAX, "set the sine frequency",),
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OPT_DBL("beep_factor", beep_factor, 0, 0, DBL_MAX, "set the beep frequency factor",),
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OPT_DBL("b", beep_factor, 0, 0, DBL_MAX, "set the beep frequency factor",),
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OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
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OPT_INT("r", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
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OPT_DUR("duration", duration, 0, 0, INT64_MAX, "set the audio duration",),
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OPT_DUR("d", duration, 0, 0, INT64_MAX, "set the audio duration",),
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OPT_STR("samples_per_frame", samples_per_frame, "1024", 0, 0, "set the number of samples per frame",),
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{NULL}
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};
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AVFILTER_DEFINE_CLASS(sine);
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#define LOG_PERIOD 15
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#define AMPLITUDE 4095
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#define AMPLITUDE_SHIFT 3
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static void make_sin_table(int16_t *sin)
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{
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unsigned half_pi = 1 << (LOG_PERIOD - 2);
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unsigned ampls = AMPLITUDE << AMPLITUDE_SHIFT;
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uint64_t unit2 = (uint64_t)(ampls * ampls) << 32;
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unsigned step, i, c, s, k, new_k, n2;
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/* Principle: if u = exp(i*a1) and v = exp(i*a2), then
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exp(i*(a1+a2)/2) = (u+v) / length(u+v) */
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sin[0] = 0;
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sin[half_pi] = ampls;
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for (step = half_pi; step > 1; step /= 2) {
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/* k = (1 << 16) * amplitude / length(u+v)
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In exact values, k is constant at a given step */
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k = 0x10000;
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for (i = 0; i < half_pi / 2; i += step) {
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s = sin[i] + sin[i + step];
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c = sin[half_pi - i] + sin[half_pi - i - step];
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n2 = s * s + c * c;
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/* Newton's method to solve n² * k² = unit² */
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while (1) {
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new_k = (k + unit2 / ((uint64_t)k * n2) + 1) >> 1;
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if (k == new_k)
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break;
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k = new_k;
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}
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sin[i + step / 2] = (k * s + 0x7FFF) >> 16;
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sin[half_pi - i - step / 2] = (k * c + 0x8000) >> 16;
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}
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}
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/* Unshift amplitude */
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for (i = 0; i <= half_pi; i++)
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sin[i] = (sin[i] + (1 << (AMPLITUDE_SHIFT - 1))) >> AMPLITUDE_SHIFT;
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/* Use symmetries to fill the other three quarters */
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for (i = 0; i < half_pi; i++)
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sin[half_pi * 2 - i] = sin[i];
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for (i = 0; i < 2 * half_pi; i++)
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sin[i + 2 * half_pi] = -sin[i];
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}
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static const char *const var_names[] = {
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"n",
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"pts",
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"t",
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"TB",
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NULL
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};
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enum {
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VAR_N,
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VAR_PTS,
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VAR_T,
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VAR_TB,
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VAR_VARS_NB
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};
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static void sampling_init(SamplingContext *c, double frequency, int sample_rate)
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{
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AVRational r;
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int r_den, max_r_den;
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max_r_den = INT_MAX / sample_rate;
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frequency = fmod(frequency, sample_rate);
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r = av_d2q(fmod(frequency, 1.0), max_r_den);
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r_den = FFMIN(r.den, max_r_den);
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c->dphi = ldexp(frequency, 32) / sample_rate;
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c->dphi_den = r_den * sample_rate;
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c->dphi_rem = round((ldexp(frequency, 32) / sample_rate - c->dphi) * c->dphi_den);
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if (c->dphi_rem >= c->dphi_den) {
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c->dphi++;
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c->dphi_rem = 0;
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}
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c->phi_rem = (-c->dphi_den - 1) / 2;
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}
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static av_always_inline void sampling_advance(SamplingContext *c)
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{
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c->phi += c->dphi;
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c->phi_rem += c->dphi_rem;
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if (c->phi_rem >= 0) {
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c->phi_rem -= c->dphi_den;
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c->phi++;
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}
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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int ret;
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SineContext *sine = ctx->priv;
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if (!(sine->sin = av_malloc(sizeof(*sine->sin) << LOG_PERIOD)))
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return AVERROR(ENOMEM);
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sampling_init(&sine->signal, sine->frequency, sine->sample_rate);
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make_sin_table(sine->sin);
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if (sine->beep_factor) {
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sine->beep_period = sine->sample_rate;
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sine->beep_length = sine->beep_period / 25;
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sampling_init(&sine->beep, sine->beep_factor * sine->frequency, sine->sample_rate);
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}
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ret = av_expr_parse(&sine->samples_per_frame_expr,
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sine->samples_per_frame, var_names,
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NULL, NULL, NULL, NULL, 0, sine);
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if (ret < 0)
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return ret;
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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SineContext *sine = ctx->priv;
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av_expr_free(sine->samples_per_frame_expr);
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sine->samples_per_frame_expr = NULL;
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av_freep(&sine->sin);
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}
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static av_cold int query_formats(const AVFilterContext *ctx,
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AVFilterFormatsConfig **cfg_in,
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AVFilterFormatsConfig **cfg_out)
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{
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const SineContext *sine = ctx->priv;
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static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
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int sample_rates[] = { sine->sample_rate, -1 };
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static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE };
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int ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts);
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if (ret < 0)
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return ret;
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ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, chlayouts);
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if (ret < 0)
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return ret;
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return ff_set_common_samplerates_from_list2(ctx, cfg_in, cfg_out, sample_rates);
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}
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static av_cold int config_props(AVFilterLink *outlink)
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{
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SineContext *sine = outlink->src->priv;
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sine->duration = av_rescale(sine->duration, sine->sample_rate, AV_TIME_BASE);
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return 0;
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}
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static int activate(AVFilterContext *ctx)
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{
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AVFilterLink *outlink = ctx->outputs[0];
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FilterLink *outl = ff_filter_link(outlink);
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SineContext *sine = ctx->priv;
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AVFrame *frame;
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double values[VAR_VARS_NB] = {
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[VAR_N] = outl->frame_count_in,
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[VAR_PTS] = sine->pts,
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[VAR_T] = sine->pts * av_q2d(outlink->time_base),
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[VAR_TB] = av_q2d(outlink->time_base),
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};
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int i, nb_samples = lrint(av_expr_eval(sine->samples_per_frame_expr, values, sine));
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int16_t *samples;
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if (!ff_outlink_frame_wanted(outlink))
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return FFERROR_NOT_READY;
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if (nb_samples <= 0) {
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av_log(sine, AV_LOG_WARNING, "nb samples expression evaluated to %d, "
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"defaulting to 1024\n", nb_samples);
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nb_samples = 1024;
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}
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if (sine->duration) {
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nb_samples = FFMIN(nb_samples, sine->duration - sine->pts);
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av_assert1(nb_samples >= 0);
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if (!nb_samples) {
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ff_outlink_set_status(outlink, AVERROR_EOF, sine->pts);
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return 0;
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}
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}
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if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
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return AVERROR(ENOMEM);
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samples = (int16_t *)frame->data[0];
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for (i = 0; i < nb_samples; i++) {
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samples[i] = sine->sin[sine->signal.phi >> (32 - LOG_PERIOD)];
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sampling_advance(&sine->signal);
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if (sine->beep_index < sine->beep_length) {
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samples[i] += sine->sin[sine->beep.phi >> (32 - LOG_PERIOD)] * 2;
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sampling_advance(&sine->beep);
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}
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if (++sine->beep_index == sine->beep_period)
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sine->beep_index = 0;
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}
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frame->pts = sine->pts;
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sine->pts += nb_samples;
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return ff_filter_frame(outlink, frame);
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}
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static const AVFilterPad sine_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_props,
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},
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};
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const AVFilter ff_asrc_sine = {
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.name = "sine",
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.description = NULL_IF_CONFIG_SMALL("Generate sine wave audio signal."),
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.init = init,
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.uninit = uninit,
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.activate = activate,
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.priv_size = sizeof(SineContext),
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.inputs = NULL,
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FILTER_OUTPUTS(sine_outputs),
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FILTER_QUERY_FUNC2(query_formats),
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.priv_class = &sine_class,
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};
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