[FFmpeg-devel] [PATCH] avfilter: add anoisesrc

Kyle Swanson k at ylo.ph
Wed Nov 4 19:19:57 CET 2015


Signed-off-by: Kyle Swanson <k at ylo.ph>
---
 Changelog                    |   1 +
 doc/filters.texi             |  36 +++++++
 libavfilter/Makefile         |   1 +
 libavfilter/allfilters.c     |   1 +
 libavfilter/asrc_anoisesrc.c | 222 +++++++++++++++++++++++++++++++++++++++++++
 libavfilter/version.h        |   4 +-
 6 files changed, 263 insertions(+), 2 deletions(-)
 create mode 100644 libavfilter/asrc_anoisesrc.c

diff --git a/Changelog b/Changelog
index 91955da..ca477de 100644
--- a/Changelog
+++ b/Changelog
@@ -30,6 +30,7 @@ version <next>:
 - innoHeim/Rsupport Screen Capture Codec decoder
 - ADPCM AICA decoder
 - Interplay ACM demuxer and audio decoder
+- anoisesrc audio source
 
 
 version 2.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index 15ea77a..620d787 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3193,6 +3193,42 @@ 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/}
 
+ at section anoisesrc
+
+Generate a noise audio signal.
+
+The filter accepts the following options:
+
+ at table @option
+
+ at item color, colour, c
+Specify the color of noise. Available noise colors are white, pink, and brown. Default color is white.
+
+ at item sample_rate, r
+Specify the sample rate. Default value is 48000 Hz.
+
+ at item duration, d
+Specify the duration of the generated audio stream. Not specifying this option results in noise with an infinite length. 
+
+ at item amplitude, a
+Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.
+
+ at item seed, s
+Specify a value used to seed the PRNG. Default value is 0.
+
+ at end table
+
+ at subsection Examples
+
+ at itemize
+
+ at item
+Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:
+ at example
+anoisesrc=d=60:c=pink:r=44100:a=0.5
+ at end example
+ at 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 1b23085..5f60e15 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-$(CONFIG_ANOISESRC_FILTER)              += asrc_anoisesrc.o
 OBJS-$(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..e716174 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(ANOISESRC,      anoisesrc,      asrc);
     REGISTER_FILTER(SINE,           sine,           asrc);
 
     REGISTER_FILTER(ANULLSINK,      anullsink,      asink);
diff --git a/libavfilter/asrc_anoisesrc.c b/libavfilter/asrc_anoisesrc.c
new file mode 100644
index 0000000..d008d67
--- /dev/null
+++ b/libavfilter/asrc_anoisesrc.c
@@ -0,0 +1,222 @@
+/*
+ * Copyright (c) 2015 Kyle Swanson <k at 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;
+    int64_t dur;
+    char *color;
+    int seed;
+
+    int infinite;
+    double (*filter)(double white, double *buf);
+    double* buf;
+    int buf_size;
+    AVLFG c;
+} ANoiseSrcContext;
+
+#define OFFSET(x) offsetof(ANoiseSrcContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption anoisesrc_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),          AV_OPT_TYPE_DURATION,  {.i64 = 0},       0,         INT64_MAX, FLAGS },
+    { "d",            "set duration",     OFFSET(dur),          AV_OPT_TYPE_DURATION,  {.i64 = 0},       0,         INT64_MAX, FLAGS },
+    { "color",        "set noise color",  OFFSET(color),        AV_OPT_TYPE_STRING,    {.str = "white"},  CHAR_MIN,  CHAR_MAX,  FLAGS },
+    { "colour",       "set noise color",  OFFSET(color),        AV_OPT_TYPE_STRING,    {.str = "white"},  CHAR_MIN,  CHAR_MAX,  FLAGS },
+    { "c",            "set noise color",  OFFSET(color),        AV_OPT_TYPE_STRING,    {.str = "white"},  CHAR_MIN,  CHAR_MAX,  FLAGS },
+    { "seed",         "set random seed",  OFFSET(seed),         AV_OPT_TYPE_INT,       {.i64 = 0},        0,         UINT_MAX,   FLAGS },
+    { "s",            "set random seed",  OFFSET(seed),         AV_OPT_TYPE_INT,       {.i64 = 0},        0,         UINT_MAX,   FLAGS },
+    {NULL}
+};
+
+AVFILTER_DEFINE_CLASS(anoisesrc);
+
+static av_cold int query_formats(AVFilterContext *ctx)
+{
+    ANoiseSrcContext *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 double white_filter(double white, double *buf) {
+    return white;
+};
+
+static double pink_filter(double white, double *buf) {
+    double pink;
+
+    /* http://www.musicdsp.org/files/pink.txt */
+    buf[0] = 0.99886 * buf[0] + white * 0.0555179;
+    buf[1] = 0.99332 * buf[1] + white * 0.0750759;
+    buf[2] = 0.96900 * buf[2] + white * 0.1538520;
+    buf[3] = 0.86650 * buf[3] + white * 0.3104856;
+    buf[4] = 0.55000 * buf[4] + white * 0.5329522;
+    buf[5] = -0.7616 * buf[5] - white * 0.0168980;
+    pink = buf[0] + buf[1] + buf[2] + buf[3] + buf[4] + buf[5] + buf[6] + white * 0.5362;
+    buf[6] = white * 0.115926;
+    return pink * 0.11;
+}
+
+static double brown_filter(double white, double *buf) {
+    double brown;
+
+    brown = ((0.02 * white) + buf[0]) / 1.02;
+    buf[0] = brown;
+    return brown * 3.5;
+}
+
+static av_cold int config_props(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    ANoiseSrcContext *s = ctx->priv;
+    if (s->dur == 0) {
+        s->infinite = 1;
+    } else {
+        s->dur = av_rescale(s->dur, s->sample_rate, AV_TIME_BASE);
+    }
+    return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx) {
+    ANoiseSrcContext *s = ctx->priv;
+
+    av_lfg_init(&s->c, s->seed);
+
+    if (!strcmp(s->color, "pink")) {
+        s->filter = pink_filter;
+        s->buf_size = 7;
+    } else if(!strcmp(s->color, "brown")) {
+        s->filter = brown_filter;
+        s->buf_size = 1;
+    } else if(!strcmp(s->color, "white")) {
+        s->filter = white_filter;
+        s->buf_size = 0;
+    } else {
+        av_log(ctx, AV_LOG_ERROR, "Invalid noise color: %s\n", s->color);
+        return AVERROR_OPTION_NOT_FOUND;
+    }
+
+    if (s->buf_size > 0) {
+        s->buf = av_malloc_array(s->buf_size, sizeof(double));
+        if (!s->buf)
+            return AVERROR(ENOMEM);
+        for (int i = 0; i < s->buf_size; i++)
+            s->buf[i] = 0;
+    }
+
+    return 0;
+}
+
+static int request_frame(AVFilterLink *outlink) {
+    AVFilterContext *ctx = outlink->src;
+    ANoiseSrcContext *s = ctx->priv;
+    AVFrame *frame;
+    int nb_samples, i;
+    double *dst;
+
+    if (!s->infinite && s->dur <= 0) {
+        return AVERROR_EOF;
+    } else if (!s->infinite && s->dur < 1024) {
+        nb_samples = s->dur;
+    } else {
+        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++) {
+        double white;
+        white = s->amplitude * ((2 * ((double) av_lfg_get(&s->c) / 0xffffffff)) - 1);
+        dst[i] = s->filter(white, s->buf);
+    }
+
+    s->dur -= nb_samples;
+    return ff_filter_frame(outlink, frame);
+}
+
+static av_cold void uninit(AVFilterContext *ctx) {
+    ANoiseSrcContext *s = ctx->priv;
+    if (s->buf_size > 0)
+        av_freep(&s->buf);
+}
+
+static const AVFilterPad anoisesrc_outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .request_frame = request_frame,
+        .config_props  = config_props,
+    },
+    { NULL }
+};
+
+AVFilter ff_asrc_anoisesrc = {
+    .name          = "anoisesrc",
+    .description   = NULL_IF_CONFIG_SMALL("Generate a noise audio signal."),
+    .init          = init,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .priv_size     = sizeof(ANoiseSrcContext),
+    .inputs        = NULL,
+    .outputs       = anoisesrc_outputs,
+    .priv_class    = &anoisesrc_class,
+};
diff --git a/libavfilter/version.h b/libavfilter/version.h
index c3ecf91..ed3b642 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,8 +30,8 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  14
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MINOR  15
+#define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \
-- 
2.6.2



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