[FFmpeg-devel] [PATCH 1/4] lavfi: remove af_resample

wm4 nfxjfg at googlemail.com
Mon Mar 6 09:51:36 EET 2017


On Mon,  6 Mar 2017 02:46:48 +0000
Rostislav Pehlivanov <atomnuker at gmail.com> wrote:

> af_aresample does the same thing better and doesn't depend on
> libavresample
> 
> Signed-off-by: Rostislav Pehlivanov <atomnuker at gmail.com>
> ---
>  libavfilter/Makefile      |   1 -
>  libavfilter/af_resample.c | 357 ----------------------------------------------
>  2 files changed, 358 deletions(-)
>  delete mode 100644 libavfilter/af_resample.c
> 
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 0ba1c74a26..6b9fba2d4c 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -98,7 +98,6 @@ OBJS-$(CONFIG_LOUDNORM_FILTER)               += af_loudnorm.o ebur128.o
>  OBJS-$(CONFIG_LOWPASS_FILTER)                += af_biquads.o
>  OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o
>  OBJS-$(CONFIG_REPLAYGAIN_FILTER)             += af_replaygain.o
> -OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
>  OBJS-$(CONFIG_RUBBERBAND_FILTER)             += af_rubberband.o
>  OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER)      += af_sidechaincompress.o
>  OBJS-$(CONFIG_SIDECHAINGATE_FILTER)          += af_agate.o
> diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c
> deleted file mode 100644
> index e3c6a20696..0000000000
> --- a/libavfilter/af_resample.c
> +++ /dev/null
> @@ -1,357 +0,0 @@
> -/*
> - * 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
> - */
> -
> -/**
> - * @file
> - * sample format and channel layout conversion audio filter
> - */
> -
> -#include "libavutil/avassert.h"
> -#include "libavutil/avstring.h"
> -#include "libavutil/common.h"
> -#include "libavutil/dict.h"
> -#include "libavutil/mathematics.h"
> -#include "libavutil/opt.h"
> -
> -#include "libavresample/avresample.h"
> -
> -#include "audio.h"
> -#include "avfilter.h"
> -#include "formats.h"
> -#include "internal.h"
> -
> -typedef struct ResampleContext {
> -    const AVClass *class;
> -    AVAudioResampleContext *avr;
> -    AVDictionary *options;
> -
> -    int resampling;
> -    int64_t next_pts;
> -    int64_t next_in_pts;
> -
> -    /* set by filter_frame() to signal an output frame to request_frame() */
> -    int got_output;
> -} ResampleContext;
> -
> -static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
> -{
> -    ResampleContext *s = ctx->priv;
> -    const AVClass *avr_class = avresample_get_class();
> -    AVDictionaryEntry *e = NULL;
> -
> -    while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
> -        if (av_opt_find(&avr_class, e->key, NULL, 0,
> -                        AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))
> -            av_dict_set(&s->options, e->key, e->value, 0);
> -    }
> -
> -    e = NULL;
> -    while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
> -        av_dict_set(opts, e->key, NULL, 0);
> -
> -    /* do not allow the user to override basic format options */
> -    av_dict_set(&s->options,  "in_channel_layout", NULL, 0);
> -    av_dict_set(&s->options, "out_channel_layout", NULL, 0);
> -    av_dict_set(&s->options,  "in_sample_fmt",     NULL, 0);
> -    av_dict_set(&s->options, "out_sample_fmt",     NULL, 0);
> -    av_dict_set(&s->options,  "in_sample_rate",    NULL, 0);
> -    av_dict_set(&s->options, "out_sample_rate",    NULL, 0);
> -
> -    return 0;
> -}
> -
> -static av_cold void uninit(AVFilterContext *ctx)
> -{
> -    ResampleContext *s = ctx->priv;
> -
> -    if (s->avr) {
> -        avresample_close(s->avr);
> -        avresample_free(&s->avr);
> -    }
> -    av_dict_free(&s->options);
> -}
> -
> -static int query_formats(AVFilterContext *ctx)
> -{
> -    AVFilterLink *inlink  = ctx->inputs[0];
> -    AVFilterLink *outlink = ctx->outputs[0];
> -    AVFilterFormats *in_formats, *out_formats, *in_samplerates, *out_samplerates;
> -    AVFilterChannelLayouts *in_layouts, *out_layouts;
> -    int ret;
> -
> -    if (!(in_formats      = ff_all_formats         (AVMEDIA_TYPE_AUDIO)) ||
> -        !(out_formats     = ff_all_formats         (AVMEDIA_TYPE_AUDIO)) ||
> -        !(in_samplerates  = ff_all_samplerates     (                  )) ||
> -        !(out_samplerates = ff_all_samplerates     (                  )) ||
> -        !(in_layouts      = ff_all_channel_layouts (                  )) ||
> -        !(out_layouts     = ff_all_channel_layouts (                  )))
> -        return AVERROR(ENOMEM);
> -
> -    if ((ret = ff_formats_ref         (in_formats,      &inlink->out_formats        )) < 0 ||
> -        (ret = ff_formats_ref         (out_formats,     &outlink->in_formats        )) < 0 ||
> -        (ret = ff_formats_ref         (in_samplerates,  &inlink->out_samplerates    )) < 0 ||
> -        (ret = ff_formats_ref         (out_samplerates, &outlink->in_samplerates    )) < 0 ||
> -        (ret = ff_channel_layouts_ref (in_layouts,      &inlink->out_channel_layouts)) < 0 ||
> -        (ret = ff_channel_layouts_ref (out_layouts,     &outlink->in_channel_layouts)) < 0)
> -        return ret;
> -
> -    return 0;
> -}
> -
> -static int config_output(AVFilterLink *outlink)
> -{
> -    AVFilterContext *ctx = outlink->src;
> -    AVFilterLink *inlink = ctx->inputs[0];
> -    ResampleContext   *s = ctx->priv;
> -    char buf1[64], buf2[64];
> -    int ret;
> -
> -    int64_t resampling_forced;
> -
> -    if (s->avr) {
> -        avresample_close(s->avr);
> -        avresample_free(&s->avr);
> -    }
> -
> -    if (inlink->channel_layout == outlink->channel_layout &&
> -        inlink->sample_rate    == outlink->sample_rate    &&
> -        (inlink->format        == outlink->format ||
> -        (av_get_channel_layout_nb_channels(inlink->channel_layout)  == 1 &&
> -         av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
> -         av_get_planar_sample_fmt(inlink->format) ==
> -         av_get_planar_sample_fmt(outlink->format))))
> -        return 0;
> -
> -    if (!(s->avr = avresample_alloc_context()))
> -        return AVERROR(ENOMEM);
> -
> -    if (s->options) {
> -        int ret;
> -        AVDictionaryEntry *e = NULL;
> -        while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
> -            av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
> -
> -        ret = av_opt_set_dict(s->avr, &s->options);
> -        if (ret < 0)
> -            return ret;
> -    }
> -
> -    av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
> -    av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
> -    av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
> -    av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
> -    av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
> -    av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
> -
> -    if ((ret = avresample_open(s->avr)) < 0)
> -        return ret;
> -
> -    av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);
> -    s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate);
> -
> -    if (s->resampling) {
> -        outlink->time_base = (AVRational){ 1, outlink->sample_rate };
> -        s->next_pts        = AV_NOPTS_VALUE;
> -        s->next_in_pts     = AV_NOPTS_VALUE;
> -    } else
> -        outlink->time_base = inlink->time_base;
> -
> -    av_get_channel_layout_string(buf1, sizeof(buf1),
> -                                 -1, inlink ->channel_layout);
> -    av_get_channel_layout_string(buf2, sizeof(buf2),
> -                                 -1, outlink->channel_layout);
> -    av_log(ctx, AV_LOG_VERBOSE,
> -           "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
> -           av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
> -           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
> -
> -    return 0;
> -}
> -
> -static int request_frame(AVFilterLink *outlink)
> -{
> -    AVFilterContext *ctx = outlink->src;
> -    ResampleContext   *s = ctx->priv;
> -    int ret = 0;
> -
> -    s->got_output = 0;
> -    while (ret >= 0 && !s->got_output)
> -        ret = ff_request_frame(ctx->inputs[0]);
> -
> -    /* flush the lavr delay buffer */
> -    if (ret == AVERROR_EOF && s->avr) {
> -        AVFrame *frame;
> -        int nb_samples = avresample_get_out_samples(s->avr, 0);
> -
> -        if (!nb_samples)
> -            return ret;
> -
> -        frame = ff_get_audio_buffer(outlink, nb_samples);
> -        if (!frame)
> -            return AVERROR(ENOMEM);
> -
> -        ret = avresample_convert(s->avr, frame->extended_data,
> -                                 frame->linesize[0], nb_samples,
> -                                 NULL, 0, 0);
> -        if (ret <= 0) {
> -            av_frame_free(&frame);
> -            return (ret == 0) ? AVERROR_EOF : ret;
> -        }
> -
> -        frame->nb_samples = ret;
> -        frame->pts = s->next_pts;
> -        return ff_filter_frame(outlink, frame);
> -    }
> -    return ret;
> -}
> -
> -static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> -{
> -    AVFilterContext  *ctx = inlink->dst;
> -    ResampleContext    *s = ctx->priv;
> -    AVFilterLink *outlink = ctx->outputs[0];
> -    int ret;
> -
> -    if (s->avr) {
> -        AVFrame *out;
> -        int delay, nb_samples;
> -
> -        /* maximum possible samples lavr can output */
> -        delay      = avresample_get_delay(s->avr);
> -        nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
> -
> -        out = ff_get_audio_buffer(outlink, nb_samples);
> -        if (!out) {
> -            ret = AVERROR(ENOMEM);
> -            goto fail;
> -        }
> -
> -        ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
> -                                 nb_samples, in->extended_data, in->linesize[0],
> -                                 in->nb_samples);
> -        if (ret <= 0) {
> -            av_frame_free(&out);
> -            if (ret < 0)
> -                goto fail;
> -        }
> -
> -        av_assert0(!avresample_available(s->avr));
> -
> -        if (s->resampling && s->next_pts == AV_NOPTS_VALUE) {
> -            if (in->pts == AV_NOPTS_VALUE) {
> -                av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
> -                       "assuming 0.\n");
> -                s->next_pts = 0;
> -            } else
> -                s->next_pts = av_rescale_q(in->pts, inlink->time_base,
> -                                           outlink->time_base);
> -        }
> -
> -        if (ret > 0) {
> -            out->nb_samples = ret;
> -
> -            ret = av_frame_copy_props(out, in);
> -            if (ret < 0) {
> -                av_frame_free(&out);
> -                goto fail;
> -            }
> -
> -            if (s->resampling) {
> -                out->sample_rate = outlink->sample_rate;
> -                /* Only convert in->pts if there is a discontinuous jump.
> -                   This ensures that out->pts tracks the number of samples actually
> -                   output by the resampler in the absence of such a jump.
> -                   Otherwise, the rounding in av_rescale_q() and av_rescale()
> -                   causes off-by-1 errors. */
> -                if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
> -                    out->pts = av_rescale_q(in->pts, inlink->time_base,
> -                                                outlink->time_base) -
> -                                   av_rescale(delay, outlink->sample_rate,
> -                                              inlink->sample_rate);
> -                } else
> -                    out->pts = s->next_pts;
> -
> -                s->next_pts = out->pts + out->nb_samples;
> -                s->next_in_pts = in->pts + in->nb_samples;
> -            } else
> -                out->pts = in->pts;
> -
> -            ret = ff_filter_frame(outlink, out);
> -            s->got_output = 1;
> -        }
> -
> -fail:
> -        av_frame_free(&in);
> -    } else {
> -        in->format = outlink->format;
> -        ret = ff_filter_frame(outlink, in);
> -        s->got_output = 1;
> -    }
> -
> -    return ret;
> -}
> -
> -static const AVClass *resample_child_class_next(const AVClass *prev)
> -{
> -    return prev ? NULL : avresample_get_class();
> -}
> -
> -static void *resample_child_next(void *obj, void *prev)
> -{
> -    ResampleContext *s = obj;
> -    return prev ? NULL : s->avr;
> -}
> -
> -static const AVClass resample_class = {
> -    .class_name       = "resample",
> -    .item_name        = av_default_item_name,
> -    .version          = LIBAVUTIL_VERSION_INT,
> -    .child_class_next = resample_child_class_next,
> -    .child_next       = resample_child_next,
> -};
> -
> -static const AVFilterPad avfilter_af_resample_inputs[] = {
> -    {
> -        .name          = "default",
> -        .type          = AVMEDIA_TYPE_AUDIO,
> -        .filter_frame  = filter_frame,
> -    },
> -    { NULL }
> -};
> -
> -static const AVFilterPad avfilter_af_resample_outputs[] = {
> -    {
> -        .name          = "default",
> -        .type          = AVMEDIA_TYPE_AUDIO,
> -        .config_props  = config_output,
> -        .request_frame = request_frame
> -    },
> -    { NULL }
> -};
> -
> -AVFilter ff_af_resample = {
> -    .name          = "resample",
> -    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
> -    .priv_size     = sizeof(ResampleContext),
> -    .priv_class    = &resample_class,
> -    .init_dict     = init,
> -    .uninit        = uninit,
> -    .query_formats = query_formats,
> -    .inputs        = avfilter_af_resample_inputs,
> -    .outputs       = avfilter_af_resample_outputs,
> -};

I'd rather remove af_aresample.c (and port af_resample.c to
libswsresample or whatever if you want), because af_resample.c has the
better filter name.


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