Hi,
On Wed, Nov 4, 2015 at 12:30 PM, Nicolas George <geo...@nsup.org> wrote:
> 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.
Yep. Will fix this.
>
>>
>> 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.
>
I can change this, but most filters I've seen have used floating point
sample formats. Anyone else have any opinions on this?
>> +
>> + 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.
>
This makes it easier for someone to add different flavors of filtered
noise later on, and define their own sample buffer. I understand this
is a tiny buffer, but why allocate too much memory if we won't need
it?
>> + 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
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Thanks for your comments. I'll wait to see if anyone else has anything
to add, and I'll send an updated patch.
Kyle
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