[FFmpeg-devel] [PATCH] lavfi: add asetrate filter.
Stefano Sabatini
stefasab at gmail.com
Fri Apr 19 17:50:07 CEST 2013
On date Thursday 2013-04-18 17:02:11 +0200, Nicolas George encoded:
>
> Signed-off-by: Nicolas George <nicolas.george at normalesup.org>
> ---
> Changelog | 1 +
> doc/filters.texi | 12 +++++
> libavfilter/Makefile | 1 +
> libavfilter/af_asetrate.c | 119 +++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> libavfilter/version.h | 4 +-
> 6 files changed, 136 insertions(+), 2 deletions(-)
> create mode 100644 libavfilter/af_asetrate.c
>
> diff --git a/Changelog b/Changelog
> index c5383ff..1d6cdce 100644
> --- a/Changelog
> +++ b/Changelog
> @@ -23,6 +23,7 @@ version <next>:
> - new interlace filter
> - smptehdbars source
> - inverse telecine filters (fieldmatch and decimate)
> +- asetrate filter
>
>
> version 1.2:
> diff --git a/doc/filters.texi b/doc/filters.texi
> index adf6000..2e30b5d 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -889,6 +889,18 @@ disable padding for the last frame, use:
> asetnsamples=n=1234:p=0
> @end example
>
> + at section asetrate
> +
> +Set the sample rate without altering the PCM data.
> +This will result in a change of speed and pitch.
> +
> +The filter accepts the following options:
> +
> + at table @option
> + at item sample_rate, r
> +Set the output sample rate. Default is 44100 Hz.
> + at end table
> +
> @section ashowinfo
>
> Show a line containing various information for each input audio frame.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 04a5b39..57f863c 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -64,6 +64,7 @@ OBJS-$(CONFIG_ASELECT_FILTER) += f_select.o
> OBJS-$(CONFIG_ASENDCMD_FILTER) += f_sendcmd.o
> OBJS-$(CONFIG_ASETNSAMPLES_FILTER) += af_asetnsamples.o
> OBJS-$(CONFIG_ASETPTS_FILTER) += f_setpts.o
> +OBJS-$(CONFIG_ASETRATE_FILTER) += af_asetrate.o
> OBJS-$(CONFIG_ASETTB_FILTER) += f_settb.o
> OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
> OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
> diff --git a/libavfilter/af_asetrate.c b/libavfilter/af_asetrate.c
> new file mode 100644
> index 0000000..e031b89
> --- /dev/null
> +++ b/libavfilter/af_asetrate.c
> @@ -0,0 +1,119 @@
> +/*
> + * Copyright (c) 2013 Nicolas George
> + *
> + * 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 "libavutil/opt.h"
> +#include "avfilter.h"
> +#include "internal.h"
> +
> +typedef struct {
> + const AVClass *class;
> + int sample_rate;
> + int rescale_pts;
> +} ASetRateContext;
> +
> +#define CONTEXT ASetRateContext
> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +#define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
> + { name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type, \
> + { .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
> +
> +#define OPT_INT(name, field, def, min, max, descr, ...) \
> + OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
> +
> +static const AVOption asetrate_options[] = {
> + OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate"),
> + OPT_INT("r", sample_rate, 44100, 1, INT_MAX, "set the sample rate"),
> + {NULL},
> +};
> +
> +AVFILTER_DEFINE_CLASS(asetrate);
> +
> +static av_cold int query_formats(AVFilterContext *ctx)
> +{
> + ASetRateContext *sr = ctx->priv;
> + int sample_rates[] = { sr->sample_rate, -1 };
> +
> + ff_formats_ref(ff_make_format_list(sample_rates),
> + &ctx->outputs[0]->in_samplerates);
> + return 0;
> +}
> +
> +static av_cold int config_props(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + ASetRateContext *sr = ctx->priv;
> + AVFilterLink *inlink = ctx->inputs[0];
> + AVRational intb = ctx->inputs[0]->time_base;
> + int inrate = inlink->sample_rate;
> +
> + if (intb.num == 1 && intb.den == inrate) {
> + outlink->time_base.num = 1;
> + outlink->time_base.den = outlink->sample_rate;
> + } else {
> + outlink->time_base = intb;
> + sr->rescale_pts = 1;
> + if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
> + av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
Is condition:
av_q2d(intb) <= 1.0 / FFMAX(inrate, outlink->sample_rate)
really sufficient for accurate rescaling?
> + }
> + return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + ASetRateContext *sr = ctx->priv;
> + AVFilterLink *outlink = ctx->outputs[0];
> +
> + frame->sample_rate = outlink->sample_rate;
> + if (sr->rescale_pts)
> + frame->pts = av_rescale(frame->pts, inlink->sample_rate,
> + outlink->sample_rate);
> + return ff_filter_frame(outlink, frame);
> +}
> +
> +static const AVFilterPad asetrate_inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad asetrate_outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .config_props = config_props,
> + },
> + { NULL }
> +};
> +
> +AVFilter avfilter_af_asetrate = {
> + .name = "asetrate",
> + .description = NULL_IF_CONFIG_SMALL("Change the sample rate without "
> + "altering the data."),
> + .query_formats = query_formats,
> + .priv_size = sizeof(ASetRateContext),
> + .inputs = asetrate_inputs,
> + .outputs = asetrate_outputs,
> + .priv_class = &asetrate_class,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 95bd270..1e3c22f 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -62,6 +62,7 @@ void avfilter_register_all(void)
> REGISTER_FILTER(ASENDCMD, asendcmd, af);
> REGISTER_FILTER(ASETNSAMPLES, asetnsamples, af);
> REGISTER_FILTER(ASETPTS, asetpts, af);
> + REGISTER_FILTER(ASETRATE, asetrate, af);
> REGISTER_FILTER(ASETTB, asettb, af);
> REGISTER_FILTER(ASHOWINFO, ashowinfo, af);
> REGISTER_FILTER(ASPLIT, asplit, af);
> diff --git a/libavfilter/version.h b/libavfilter/version.h
> index ba1ec0b..c751674 100644
> --- a/libavfilter/version.h
> +++ b/libavfilter/version.h
> @@ -29,8 +29,8 @@
> #include "libavutil/avutil.h"
>
> #define LIBAVFILTER_VERSION_MAJOR 3
> -#define LIBAVFILTER_VERSION_MINOR 56
> -#define LIBAVFILTER_VERSION_MICRO 103
> +#define LIBAVFILTER_VERSION_MINOR 57
> +#define LIBAVFILTER_VERSION_MICRO 100
>
> #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
> LIBAVFILTER_VERSION_MINOR, \
LGTM, thanks.
--
FFmpeg = Faithless Fundamental Murdering Purposeless Entertaining Game
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