Thanks for the updated patch, see comments below.
Le quartidi 14 brumaire, an CCXXIV, Kyle Swanson a écrit :
> Signed-off-by: Kyle Swanson <k...@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/}
>
> +@section anoisesrc
> +
> +Generate a noise audio signal.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +
> +@item color, colour, c
> +Specify the color of noise. Available noise colors are white, pink, and
> brown. Default color is white.
> +
> +@item sample_rate, r
> +Specify the sample rate. Default value is 48000 Hz.
> +
> +@item duration, d
> +Specify the duration of the generated audio stream. Not specifying this
> option results in noise with an infinite length.
> +
> +@item amplitude, a
> +Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default
> value is 1.0.
> +
> +@item seed, s
> +Specify a value used to seed the PRNG. Default value is 0.
> +
> +@end table
> +
> +@subsection Examples
> +
> +@itemize
> +
> +@item
> +Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an
> amplitude of 0.5:
> +@example
> +anoisesrc=d=60:c=pink: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 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
Alphabetic order after renaming the filter.
>
> 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...@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
> + };
I already commented on that: please avoid floating-point computations unless
they are absolutely necessary.
> +
> + 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;
> + }
Better use AV_OPT_TYPE_FLAG for that.
> +
> + if (s->buf_size > 0) {
> + s->buf = av_malloc_array(s->buf_size, sizeof(double));
Unless I am mistaken, buf_size will be at most 7. I do not think allocating
it dynamically is worth it, just allocate it directly in the structure.
> + 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, \
Regards,
--
Nicolas George
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