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

[Paul B Mahol]

On 4/19/19, Carl Eugen Hoyos <ceffmpeg at gmail.com> wrote:
> 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?

This is not same as resampling.

This is used as part of other filtering. Namely when filter needs to oversample
audio when processing - it then upsample audio before processing and downsample
it by same factor after processing it.