On Mon, 6 Mar 2017 02:46:48 +0000
Rostislav Pehlivanov <atomnu...@gmail.com> wrote:
> af_aresample does the same thing better and doesn't depend on
> libavresample
>
> Signed-off-by: Rostislav Pehlivanov <atomnu...@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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