Signed-off-by: Kyle Swanson <k...@ylo.ph> --- Changelog | 1 + doc/filters.texi | 30 ++++++++++ libavfilter/Makefile | 1 + libavfilter/allfilters.c | 1 + libavfilter/asrc_noise.c | 141 +++++++++++++++++++++++++++++++++++++++++++++++ libavfilter/version.h | 2 +- 6 files changed, 175 insertions(+), 1 deletion(-) create mode 100644 libavfilter/asrc_noise.c
diff --git a/Changelog b/Changelog index c49e383..d2ea2e1 100644 --- a/Changelog +++ b/Changelog @@ -29,6 +29,7 @@ version <next>: - vibrato filter - innoHeim/Rsupport Screen Capture Codec decoder - ADPCM AICA decoder +- noise audio source version 2.8: diff --git a/doc/filters.texi b/doc/filters.texi index 15ea77a..0d901cc 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -3193,6 +3193,36 @@ ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.' For more information about libflite, check: @url{http://www.speech.cs.cmu.edu/flite/} +@section noise + +Generate a white noise audio signal. + +The filter accepts the following options: + +@table @option + +@item sample_rate, r +Specify the sample rate. Default value is 48000 Hz. + +@item duration, d +Specify the duration of the generated audio stream. Default value is 10 seconds. + +@item amplitude, a +Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0. + +@end table + +@subsection Examples + +@itemize + +@item +Generate 60 seconds of white noise, with a 44.1 kHz sampling rate and an amplitude of 0.5: +@example +noise=d=60:r=44100:a=0.5 +@end example +@end itemize + @section sine Generate an audio signal made of a sine wave with amplitude 1/8. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index c5819d4..a0b2232 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -93,6 +93,7 @@ OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o OBJS-$(CONFIG_AEVALSRC_FILTER) += aeval.o OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o OBJS-$(CONFIG_FLITE_FILTER) += asrc_flite.o +OBJS-$(CONFIG_NOISE_FILTER) += asrc_noise.o OBJS-$(CONFIG_SINE_FILTER) += asrc_sine.o OBJS-$(CONFIG_ANULLSINK_FILTER) += asink_anullsink.o diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index a538b81..f820441 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -115,6 +115,7 @@ void avfilter_register_all(void) REGISTER_FILTER(AEVALSRC, aevalsrc, asrc); REGISTER_FILTER(ANULLSRC, anullsrc, asrc); REGISTER_FILTER(FLITE, flite, asrc); + REGISTER_FILTER(NOISE, noise, asrc); REGISTER_FILTER(SINE, sine, asrc); REGISTER_FILTER(ANULLSINK, anullsink, asink); diff --git a/libavfilter/asrc_noise.c b/libavfilter/asrc_noise.c new file mode 100644 index 0000000..00370a6 --- /dev/null +++ b/libavfilter/asrc_noise.c @@ -0,0 +1,141 @@ +/* + * Copyright (c) 2015 Kyle Swanson <k...@ylo.ph>. + * + * 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/opt.h" +#include "audio.h" +#include "avfilter.h" +#include "internal.h" +#include "libavutil/lfg.h" + +typedef struct { + const AVClass *class; + int sample_rate; + double amplitude; + double dur_sec; + int64_t dur_samp; + AVLFG c; +} NoiseContext; + +#define OFFSET(x) offsetof(NoiseContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption noise_options[] = { + { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 15, INT_MAX, FLAGS }, + { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 15, INT_MAX, FLAGS }, + { "amplitude", "set amplitude", OFFSET(amplitude), AV_OPT_TYPE_DOUBLE, {.dbl = 1.}, 0., 1., FLAGS }, + { "a", "set amplitude", OFFSET(amplitude), AV_OPT_TYPE_DOUBLE, {.dbl = 1.}, 0., 1., FLAGS }, + { "duration", "set duration", OFFSET(dur_sec), AV_OPT_TYPE_DOUBLE, {.dbl = 10.}, 0., DBL_MAX, FLAGS }, + { "d", "set duration", OFFSET(dur_sec), AV_OPT_TYPE_DOUBLE, {.dbl = 10.}, 0., DBL_MAX, FLAGS }, + {NULL} +}; + +AVFILTER_DEFINE_CLASS(noise); + +static av_cold int query_formats(AVFilterContext *ctx) +{ + NoiseContext *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_DBL, + AV_SAMPLE_FMT_NONE + }; + + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + int ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats (ctx, formats); + if (ret < 0) + return ret; + + layouts = avfilter_make_format64_list(chlayouts); + if (!layouts) + return AVERROR(ENOMEM); + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return ret; + + formats = ff_make_format_list(sample_rates); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static av_cold int config_props(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + NoiseContext *s = ctx->priv; + s->dur_samp = s->dur_sec * s->sample_rate; + return 0; +} + +static int request_frame(AVFilterLink *outlink) { + AVFilterContext *ctx = outlink->src; + NoiseContext *s = ctx->priv; + AVFrame *frame; + int nb_samples, i; + double *dst; + + nb_samples = 1024; + if (!(frame = ff_get_audio_buffer(outlink, nb_samples))) + return AVERROR(ENOMEM); + dst = (double *)frame->data[0]; + + for (i = 0; i < nb_samples; i++) { + dst[i]= s->amplitude * ((2 * ((double) av_lfg_get(&s->c) / 0xffffffff)) - 1); + s->dur_samp--; + if (s->dur_samp <= 0) + return AVERROR_EOF; + } + return ff_filter_frame(outlink, frame); +} + +static av_cold int init(AVFilterContext *ctx) { + NoiseContext *s = ctx->priv; + av_lfg_init(&s->c, 0); + return 0; +} + +static const AVFilterPad noise_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .request_frame = request_frame, + .config_props = config_props, + }, + { NULL } +}; + +AVFilter ff_asrc_noise = { + .name = "noise", + .description = NULL_IF_CONFIG_SMALL("Generate white noise audio signal."), + .init = init, + .query_formats = query_formats, + .priv_size = sizeof(NoiseContext), + .inputs = NULL, + .outputs = noise_outputs, + .priv_class = &noise_class, +}; diff --git a/libavfilter/version.h b/libavfilter/version.h index c3ecf91..05b0735 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 6 -#define LIBAVFILTER_VERSION_MINOR 14 +#define LIBAVFILTER_VERSION_MINOR 15 #define LIBAVFILTER_VERSION_MICRO 101 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ -- 2.6.2 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel