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avfilter: add anoisesrc
Signed-off-by: Kyle Swanson <k@ylo.ph> Signed-off-by: Paul B Mahol <onemda@gmail.com>
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@ -32,6 +32,7 @@ version <next>:
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- Interplay ACM demuxer and audio decoder
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- XMA1 & XMA2 decoder
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- realtime filter
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- anoisesrc audio filter source
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version 2.8:
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@ -3157,6 +3157,46 @@ 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 anoisesrc
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Generate a noise audio signal.
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The filter accepts the following options:
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@table @option
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@item sample_rate, r
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Specify the sample rate. Default value is 48000 Hz.
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@item amplitude, a
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Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value
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is 1.0.
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@item duration, d
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Specify the duration of the generated audio stream. Not specifying this option
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results in noise with an infinite length.
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@item color, colour, c
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Specify the color of noise. Available noise colors are white, pink, and brown.
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Default color is white.
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@item seed, s
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Specify a value used to seed the PRNG.
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@item nb_samples, n
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Set the number of samples per each 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 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:
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@example
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anoisesrc=d=60:c=pink:r=44100:a=0.5
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@end example
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@end itemize
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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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@ -91,6 +91,7 @@ OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
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OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
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OBJS-$(CONFIG_AEVALSRC_FILTER) += aeval.o
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OBJS-$(CONFIG_ANOISESRC_FILTER) += asrc_anoisesrc.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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@ -113,6 +113,7 @@ void avfilter_register_all(void)
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REGISTER_FILTER(VOLUMEDETECT, volumedetect, af);
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REGISTER_FILTER(AEVALSRC, aevalsrc, asrc);
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REGISTER_FILTER(ANOISESRC, anoisesrc, 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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207
libavfilter/asrc_anoisesrc.c
Normal file
207
libavfilter/asrc_anoisesrc.c
Normal file
@ -0,0 +1,207 @@
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/*
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* Copyright (c) 2015 Kyle Swanson <k@ylo.ph>.
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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/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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#include "libavutil/lfg.h"
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#include "libavutil/random_seed.h"
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typedef struct {
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const AVClass *class;
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int sample_rate;
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double amplitude;
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int64_t duration;
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int64_t color;
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int64_t seed;
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int nb_samples;
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int64_t pts;
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int infinite;
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double (*filter)(double white, double *buf);
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double buf[7];
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AVLFG c;
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} ANoiseSrcContext;
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#define OFFSET(x) offsetof(ANoiseSrcContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption anoisesrc_options[] = {
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{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 15, INT_MAX, FLAGS },
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{ "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 15, INT_MAX, FLAGS },
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{ "amplitude", "set amplitude", OFFSET(amplitude), AV_OPT_TYPE_DOUBLE, {.dbl = 1.}, 0., 1., FLAGS },
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{ "a", "set amplitude", OFFSET(amplitude), AV_OPT_TYPE_DOUBLE, {.dbl = 1.}, 0., 1., FLAGS },
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{ "duration", "set duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0}, 0, INT64_MAX, FLAGS },
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{ "d", "set duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0}, 0, INT64_MAX, FLAGS },
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{ "color", "set noise color", OFFSET(color), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 2, FLAGS, "color" },
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{ "colour", "set noise color", OFFSET(color), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 2, FLAGS, "color" },
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{ "c", "set noise color", OFFSET(color), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 2, FLAGS, "color" },
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{ "white", 0, 0, AV_OPT_TYPE_CONST, {.i64 = 0}, 0, 0, FLAGS, "color" },
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{ "pink", 0, 0, AV_OPT_TYPE_CONST, {.i64 = 1}, 0, 0, FLAGS, "color" },
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{ "brown", 0, 0, AV_OPT_TYPE_CONST, {.i64 = 2}, 0, 0, FLAGS, "color" },
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{ "seed", "set random seed", OFFSET(seed), AV_OPT_TYPE_INT64, {.i64 = -1}, -1, UINT_MAX, FLAGS },
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{ "s", "set random seed", OFFSET(seed), AV_OPT_TYPE_INT64, {.i64 = -1}, -1, UINT_MAX, FLAGS },
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{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
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{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
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{NULL}
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};
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AVFILTER_DEFINE_CLASS(anoisesrc);
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static av_cold int query_formats(AVFilterContext *ctx)
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{
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ANoiseSrcContext *s = ctx->priv;
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static const int64_t chlayouts[] = { AV_CH_LAYOUT_MONO, -1 };
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int sample_rates[] = { s->sample_rate, -1 };
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_NONE
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};
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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int ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats (ctx, formats);
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if (ret < 0)
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return ret;
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layouts = avfilter_make_format64_list(chlayouts);
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_rates);
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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static double white_filter(double white, double *buf)
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{
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return white;
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};
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static double pink_filter(double white, double *buf)
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{
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double pink;
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/* http://www.musicdsp.org/files/pink.txt */
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buf[0] = 0.99886 * buf[0] + white * 0.0555179;
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buf[1] = 0.99332 * buf[1] + white * 0.0750759;
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buf[2] = 0.96900 * buf[2] + white * 0.1538520;
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buf[3] = 0.86650 * buf[3] + white * 0.3104856;
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buf[4] = 0.55000 * buf[4] + white * 0.5329522;
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buf[5] = -0.7616 * buf[5] - white * 0.0168980;
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pink = buf[0] + buf[1] + buf[2] + buf[3] + buf[4] + buf[5] + buf[6] + white * 0.5362;
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buf[6] = white * 0.115926;
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return pink * 0.11;
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}
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static double brown_filter(double white, double *buf)
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{
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double brown;
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brown = ((0.02 * white) + buf[0]) / 1.02;
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buf[0] = brown;
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return brown * 3.5;
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}
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static av_cold int config_props(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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ANoiseSrcContext *s = ctx->priv;
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if (s->seed == -1)
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s->seed = av_get_random_seed();
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av_lfg_init(&s->c, s->seed);
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if (s->duration == 0)
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s->infinite = 1;
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s->duration = av_rescale(s->duration, s->sample_rate, AV_TIME_BASE);
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switch (s->color) {
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case 0: s->filter = white_filter; break;
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case 1: s->filter = pink_filter; break;
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case 2: s->filter = brown_filter; break;
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}
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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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AVFilterContext *ctx = outlink->src;
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ANoiseSrcContext *s = ctx->priv;
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AVFrame *frame;
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int nb_samples, i;
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double *dst;
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if (!s->infinite && s->duration <= 0) {
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return AVERROR_EOF;
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} else if (!s->infinite && s->duration < s->nb_samples) {
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nb_samples = s->duration;
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} else {
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nb_samples = s->nb_samples;
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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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dst = (double *)frame->data[0];
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for (i = 0; i < nb_samples; i++) {
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double white;
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white = s->amplitude * ((2 * ((double) av_lfg_get(&s->c) / 0xffffffff)) - 1);
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dst[i] = s->filter(white, s->buf);
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}
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if (!s->infinite)
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s->duration -= nb_samples;
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frame->pts = s->pts;
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s->pts += nb_samples;
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return ff_filter_frame(outlink, frame);
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}
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static const AVFilterPad anoisesrc_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 ff_asrc_anoisesrc = {
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.name = "anoisesrc",
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.description = NULL_IF_CONFIG_SMALL("Generate a noise audio signal."),
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.query_formats = query_formats,
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.priv_size = sizeof(ANoiseSrcContext),
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.inputs = NULL,
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.outputs = anoisesrc_outputs,
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.priv_class = &anoisesrc_class,
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};
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@ -30,8 +30,8 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 6
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#define LIBAVFILTER_VERSION_MINOR 14
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#define LIBAVFILTER_VERSION_MICRO 101
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#define LIBAVFILTER_VERSION_MINOR 15
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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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