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

James Almer jamrial at gmail.com
Tue Mar 7 00:45:32 EET 2017


On 3/5/2017 11:46 PM, Rostislav Pehlivanov wrote:
> af_aresample does the same thing better and doesn't depend on
> libavresample

But it depends on libswresample. What about the builds that are using
lavr but nor lswr?

Is the purpose of this set to pave the way for a lavr removal? Because
one thing is dropping ABI compatibility with libav since it was being a
pain in the ass and probably not even working, but another is dropping
whole modules or being increasingly API incompatible.

If it gets in the way of some addition or refactoring then sure, I'm ok
with an eventual removal, but if it's just "Redundant filter/library I
don't want around" then not so much, because said redundancy was accepted
in the first place to make downstream projects' and users' lives easier.

> 
> 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,
> -};
> 



More information about the ffmpeg-devel mailing list