[FFmpeg-devel] [PATCH 2/3] avfilter: add audio upsample filter

[Carl Eugen Hoyos]

2019-04-18 23:17 GMT+02:00, Paul B Mahol <onemda at gmail.com>:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
>  libavfilter/Makefile       |   1 +
>  libavfilter/af_aupsample.c | 159 +++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c   |   1 +
>  3 files changed, 161 insertions(+)
>  create mode 100644 libavfilter/af_aupsample.c
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 682df45ef5..a38bc35231 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -86,6 +86,7 @@ OBJS-$(CONFIG_ASTATS_FILTER)                 +=
> af_astats.o
>  OBJS-$(CONFIG_ASTREAMSELECT_FILTER)          += f_streamselect.o
> framesync.o
>  OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
>  OBJS-$(CONFIG_ATRIM_FILTER)                  += trim.o
> +OBJS-$(CONFIG_AUPSAMPLE_FILTER)              += af_aupsample.o
>  OBJS-$(CONFIG_AZMQ_FILTER)                   += f_zmq.o
>  OBJS-$(CONFIG_BANDPASS_FILTER)               += af_biquads.o
>  OBJS-$(CONFIG_BANDREJECT_FILTER)             += af_biquads.o
> diff --git a/libavfilter/af_aupsample.c b/libavfilter/af_aupsample.c
> new file mode 100644
> index 0000000000..ee35b9c0c6
> --- /dev/null
> +++ b/libavfilter/af_aupsample.c
> @@ -0,0 +1,159 @@
> +/*
> + * Copyright (c) 2019 Paul B Mahol
> + *
> + * 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 "libavutil/samplefmt.h"
> +#include "avfilter.h"
> +#include "audio.h"
> +#include "filters.h"
> +#include "internal.h"
> +
> +typedef struct AudioUpSampleContext {
> +    const AVClass *class;
> +    int factor;
> +
> +    int64_t next_pts;
> +} AudioUpSampleContext;
> +
> +#define OFFSET(x) offsetof(AudioUpSampleContext, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption aupsample_options[] = {
> +    { "factor", "set upsampling factor", OFFSET(factor), AV_OPT_TYPE_INT,
> {.i64=1}, 1, 64, A },
> +    { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(aupsample);
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> +    AudioUpSampleContext *s = ctx->priv;
> +    AVFilterChannelLayouts *layouts;
> +    AVFilterFormats *formats;
> +    int sample_rates[] = { 44100, -1 };
> +    static const enum AVSampleFormat sample_fmts[] = {
> +        AV_SAMPLE_FMT_DBLP,
> +        AV_SAMPLE_FMT_NONE
> +    };
> +    AVFilterFormats *avff;
> +    int ret;
> +
> +    if (!ctx->inputs[0]->in_samplerates ||
> +        !ctx->inputs[0]->in_samplerates->nb_formats) {
> +        return AVERROR(EAGAIN);
> +    }
> +
> +    layouts = ff_all_channel_counts();
> +    if (!layouts)
> +        return AVERROR(ENOMEM);
> +    ret = ff_set_common_channel_layouts(ctx, layouts);
> +    if (ret < 0)
> +        return 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;
> +
> +    avff = ctx->inputs[0]->in_samplerates;
> +    sample_rates[0] = avff->formats[0];
> +    if (!ctx->inputs[0]->out_samplerates)
> +        if ((ret = ff_formats_ref(ff_make_format_list(sample_rates),
> +                                  &ctx->inputs[0]->out_samplerates)) < 0)
> +            return ret;
> +
> +    sample_rates[0] = avff->formats[0] * s->factor;
> +    return ff_formats_ref(ff_make_format_list(sample_rates),
> +                         &ctx->outputs[0]->in_samplerates);
> +}
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    AudioUpSampleContext *s = ctx->priv;
> +
> +    s->next_pts = AV_NOPTS_VALUE;
> +
> +    return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    AudioUpSampleContext *s = ctx->priv;
> +    const int factor = s->factor;
> +    AVFrame *out;
> +
> +    if (s->factor == 1)
> +        return ff_filter_frame(outlink, in);
> +
> +    out = ff_get_audio_buffer(outlink, in->nb_samples * s->factor);
> +    if (!out) {
> +        av_frame_free(&in);
> +        return AVERROR(ENOMEM);
> +    }
> +
> +    if (s->next_pts == AV_NOPTS_VALUE)
> +        s->next_pts = in->pts;
> +
> +    for (int c = 0; c < in->channels; c++) {
> +        const double *src = (const double *)in->extended_data[c];
> +        double *dst = (double *)out->extended_data[c];
> +
> +        for (int n = 0; n < in->nb_samples; n++)
> +            dst[n*factor] = src[n];
> +    }
> +
> +    out->pts = s->next_pts;
> +    s->next_pts += av_rescale_q(out->nb_samples, (AVRational){1,
> outlink->sample_rate}, outlink->time_base);
> +    av_frame_free(&in);
> +    return ff_filter_frame(ctx->outputs[0], out);
> +}
> +
> +static const AVFilterPad aupsample_inputs[] = {
> +    {
> +        .name         = "default",
> +        .type         = AVMEDIA_TYPE_AUDIO,
> +        .filter_frame = filter_frame,
> +        .config_props = config_input,
> +    },
> +    { NULL }
> +};
> +
> +static const AVFilterPad aupsample_outputs[] = {
> +    {
> +        .name         = "default",
> +        .type         = AVMEDIA_TYPE_AUDIO,
> +    },
> +    { NULL }
> +};
> +
> +AVFilter ff_af_aupsample = {
> +    .name          = "aupsample",
> +    .description   = NULL_IF_CONFIG_SMALL("Upsample
> audio by integer factor."),

Is it faster?
Better quality?

Just wondering, Carl Eugen