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,
-};
--
2.12.0.rc1.440.g5b76565f74
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