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lavfi: add sine audio source.
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@ -10,6 +10,7 @@ version <next>:
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- perms and aperms filters
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- audio filtering support in ffplay
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- 10% faster aac encoding on x86 and MIPS
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- sine audio filter source
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version 1.2:
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@ -1653,6 +1653,57 @@ ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'
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For more information about libflite, check:
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@url{http://www.speech.cs.cmu.edu/flite/}
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@section sine
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Generate an audio signal made of a sine wave with amplitude 1/8.
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The audio signal is bit-exact.
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It accepts a list of options in the form of @var{key}=@var{value} pairs
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separated by ":". If the option name is omitted, the first option is the
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frequency and the second option is the beep factor.
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The supported options are:
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@table @option
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@item frequency, f
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Set the carrier frequency. Default is 440 Hz.
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@item beep_factor, b
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Enable a periodic beep every second with frequency @var{beep_factor} times
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the carrier frequency. Default is 0, meaning the beep is disabled.
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@item sample_rate, s
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Specify the sample rate, default is 44100.
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@item duration, d
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Specify the duration of the generated audio stream.
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@item samples_per_frame
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Set the number of samples per output frame, default is 1024.
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@end table
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@subsection Examples
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@itemize
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@item
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Generate a simple 440 Hz sine wave:
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@example
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sine
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@end example
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@item
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Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds:
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@example
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sine=220:4:d=5
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sine=f=220:b=4:d=5
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sine=frequency=220:beep_factor=4:duration=5
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@end example
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@end itemize
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@c man end AUDIO SOURCES
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@chapter Audio Sinks
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@ -90,6 +90,7 @@ OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
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OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o
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OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
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OBJS-$(CONFIG_FLITE_FILTER) += asrc_flite.o
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OBJS-$(CONFIG_SINE_FILTER) += asrc_sine.o
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OBJS-$(CONFIG_ANULLSINK_FILTER) += asink_anullsink.o
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@ -86,6 +86,7 @@ void avfilter_register_all(void)
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REGISTER_FILTER(AEVALSRC, aevalsrc, asrc);
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REGISTER_FILTER(ANULLSRC, anullsrc, asrc);
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REGISTER_FILTER(FLITE, flite, asrc);
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REGISTER_FILTER(SINE, sine, asrc);
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REGISTER_FILTER(ANULLSINK, anullsink, asink);
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228
libavfilter/asrc_sine.c
Normal file
228
libavfilter/asrc_sine.c
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@ -0,0 +1,228 @@
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/*
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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 "libavutil/avassert.h"
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#include "libavutil/channel_layout.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 "internal.h"
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typedef struct {
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const AVClass *class;
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double frequency;
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double beep_factor;
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int samples_per_frame;
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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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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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unsigned beep_period;
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unsigned beep_index;
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unsigned beep_length;
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uint32_t phi_beep; ///< current phase of the beep
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uint32_t dphi_beep; ///< phase increment of the beep
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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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static const AVOption sine_options[] = {
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OPT_DBL("frequency", frequency, 440, 0, INFINITY, "set the sine frequency"),
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OPT_DBL("f", frequency, 440, 0, INFINITY, "set the sine frequency"),
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OPT_DBL("beep_factor", beep_factor, 0, 0, INFINITY, "set the beep fequency factor"),
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OPT_DBL("b", beep_factor, 0, 0, INFINITY, "set the beep fequency 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_INT("samples_per_frame", samples_per_frame, 1024, 0, INT_MAX, "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 av_cold int init(AVFilterContext *ctx, const char *args)
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{
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SineContext *sine = ctx->priv;
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static const char *shorthand[] = { "frequency", "beep_factor", NULL };
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int ret;
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sine->class = &sine_class;
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av_opt_set_defaults(sine);
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if ((ret = av_opt_set_from_string(sine, args, shorthand, "=", ":")) < 0)
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return ret;
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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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sine->dphi = ldexp(sine->frequency, 32) / sine->sample_rate + 0.5;
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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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sine->dphi_beep = ldexp(sine->beep_factor * sine->frequency, 32) /
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sine->sample_rate + 0.5;
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}
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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_freep(&sine->sin);
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}
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static av_cold int query_formats(AVFilterContext *ctx)
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{
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SineContext *sine = ctx->priv;
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static const int64_t chlayouts[] = { AV_CH_LAYOUT_MONO, -1 };
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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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ff_set_common_formats (ctx, ff_make_format_list(sample_fmts));
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ff_set_common_channel_layouts(ctx, avfilter_make_format64_list(chlayouts));
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ff_set_common_samplerates(ctx, ff_make_format_list(sample_rates));
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return 0;
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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 request_frame(AVFilterLink *outlink)
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{
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SineContext *sine = outlink->src->priv;
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AVFrame *frame;
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int i, nb_samples = sine->samples_per_frame;
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int16_t *samples;
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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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return AVERROR_EOF;
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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->phi >> (32 - LOG_PERIOD)];
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sine->phi += sine->dphi;
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if (sine->beep_index < sine->beep_length) {
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samples[i] += sine->sin[sine->phi_beep >> (32 - LOG_PERIOD)] << 1;
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sine->phi_beep += sine->dphi_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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.request_frame = request_frame,
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.config_props = config_props,
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},
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{ NULL }
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};
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AVFilter avfilter_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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.query_formats = query_formats,
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.init = init,
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.uninit = uninit,
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.priv_size = sizeof(SineContext),
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.inputs = NULL,
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.outputs = sine_outputs,
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.priv_class = &sine_class,
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};
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@ -29,8 +29,8 @@
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#include "libavutil/avutil.h"
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#define LIBAVFILTER_VERSION_MAJOR 3
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#define LIBAVFILTER_VERSION_MINOR 47
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#define LIBAVFILTER_VERSION_MICRO 104
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#define LIBAVFILTER_VERSION_MINOR 48
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#define LIBAVFILTER_VERSION_MICRO 100
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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LIBAVFILTER_VERSION_MINOR, \
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