lavfi: remove af_asynts filter

Long overdue for removal, af_aresample should be used instead.

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>

4 files changed, 1 insertions, 326 deletions

diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index a48ca0ab45..9c15ed62d2 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -67,7 +67,6 @@ OBJS-$(CONFIG_ASIDEDATA_FILTER)              += f_sidedata.o
 OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o
 OBJS-$(CONFIG_ASTATS_FILTER)                 += af_astats.o
 OBJS-$(CONFIG_ASTREAMSELECT_FILTER)          += f_streamselect.o
-OBJS-$(CONFIG_ASYNCTS_FILTER)                += af_asyncts.o
 OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
 OBJS-$(CONFIG_ATRIM_FILTER)                  += trim.o
 OBJS-$(CONFIG_AZMQ_FILTER)                   += f_zmq.o
diff --git a/libavfilter/af_asyncts.c b/libavfilter/af_asyncts.c
deleted file mode 100644
index a33e0dd67e..0000000000
--- a/libavfilter/af_asyncts.c
+++ /dev/null
@@ -1,323 +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
- */
-
-#include <stdint.h>
-
-#include "libavresample/avresample.h"
-#include "libavutil/attributes.h"
-#include "libavutil/audio_fifo.h"
-#include "libavutil/common.h"
-#include "libavutil/mathematics.h"
-#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-
-#include "audio.h"
-#include "avfilter.h"
-#include "internal.h"
-
-typedef struct ASyncContext {
-    const AVClass *class;
-
-    AVAudioResampleContext *avr;
-    int64_t pts;            ///< timestamp in samples of the first sample in fifo
-    int min_delta;          ///< pad/trim min threshold in samples
-    int first_frame;        ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE
-    int64_t first_pts;      ///< user-specified first expected pts, in samples
-    int comp;               ///< current resample compensation
-
-    /* options */
-    int resample;
-    float min_delta_sec;
-    int max_comp;
-
-    /* set by filter_frame() to signal an output frame to request_frame() */
-    int got_output;
-} ASyncContext;
-
-#define OFFSET(x) offsetof(ASyncContext, x)
-#define A AV_OPT_FLAG_AUDIO_PARAM
-#define F AV_OPT_FLAG_FILTERING_PARAM
-static const AVOption asyncts_options[] = {
-    { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample),      AV_OPT_TYPE_BOOL,  { .i64 = 0 },   0, 1,       A|F },
-    { "min_delta",  "Minimum difference between timestamps and audio data "
-                    "(in seconds) to trigger padding/trimmin the data.",        OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A|F },
-    { "max_comp",   "Maximum compensation in samples per second.",              OFFSET(max_comp),      AV_OPT_TYPE_INT,   { .i64 = 500 }, 0, INT_MAX, A|F },
-    { "first_pts",  "Assume the first pts should be this value.",               OFFSET(first_pts),     AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A|F },
-    { NULL }
-};
-
-AVFILTER_DEFINE_CLASS(asyncts);
-
-static av_cold int init(AVFilterContext *ctx)
-{
-    ASyncContext *s = ctx->priv;
-
-    s->pts         = AV_NOPTS_VALUE;
-    s->first_frame = 1;
-
-    return 0;
-}
-
-static av_cold void uninit(AVFilterContext *ctx)
-{
-    ASyncContext *s = ctx->priv;
-
-    if (s->avr) {
-        avresample_close(s->avr);
-        avresample_free(&s->avr);
-    }
-}
-
-static int config_props(AVFilterLink *link)
-{
-    ASyncContext *s = link->src->priv;
-    int ret;
-
-    s->min_delta = s->min_delta_sec * link->sample_rate;
-    link->time_base = (AVRational){1, link->sample_rate};
-
-    s->avr = avresample_alloc_context();
-    if (!s->avr)
-        return AVERROR(ENOMEM);
-
-    av_opt_set_int(s->avr,  "in_channel_layout", link->channel_layout, 0);
-    av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0);
-    av_opt_set_int(s->avr,  "in_sample_fmt",     link->format,         0);
-    av_opt_set_int(s->avr, "out_sample_fmt",     link->format,         0);
-    av_opt_set_int(s->avr,  "in_sample_rate",    link->sample_rate,    0);
-    av_opt_set_int(s->avr, "out_sample_rate",    link->sample_rate,    0);
-
-    if (s->resample)
-        av_opt_set_int(s->avr, "force_resampling", 1, 0);
-
-    if ((ret = avresample_open(s->avr)) < 0)
-        return ret;
-
-    return 0;
-}
-
-/* get amount of data currently buffered, in samples */
-static int64_t get_delay(ASyncContext *s)
-{
-    return avresample_available(s->avr) + avresample_get_delay(s->avr);
-}
-
-static void handle_trimming(AVFilterContext *ctx)
-{
-    ASyncContext *s = ctx->priv;
-
-    if (s->pts < s->first_pts) {
-        int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr));
-        av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n",
-               delta);
-        avresample_read(s->avr, NULL, delta);
-        s->pts += delta;
-    } else if (s->first_frame)
-        s->pts = s->first_pts;
-}
-
-static int request_frame(AVFilterLink *link)
-{
-    AVFilterContext *ctx = link->src;
-    ASyncContext      *s = ctx->priv;
-    int ret = 0;
-    int nb_samples;
-
-    s->got_output = 0;
-    ret = ff_request_frame(ctx->inputs[0]);
-
-    /* flush the fifo */
-    if (ret == AVERROR_EOF) {
-        if (s->first_pts != AV_NOPTS_VALUE)
-            handle_trimming(ctx);
-
-        if (nb_samples = get_delay(s)) {
-            AVFrame *buf = ff_get_audio_buffer(link, nb_samples);
-            if (!buf)
-                return AVERROR(ENOMEM);
-            ret = avresample_convert(s->avr, buf->extended_data,
-                                     buf->linesize[0], nb_samples, NULL, 0, 0);
-            if (ret <= 0) {
-                av_frame_free(&buf);
-                return (ret < 0) ? ret : AVERROR_EOF;
-            }
-
-            buf->pts = s->pts;
-            return ff_filter_frame(link, buf);
-        }
-    }
-
-    return ret;
-}
-
-static int write_to_fifo(ASyncContext *s, AVFrame *buf)
-{
-    int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
-                                 buf->linesize[0], buf->nb_samples);
-    av_frame_free(&buf);
-    return ret;
-}
-
-static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
-{
-    AVFilterContext  *ctx = inlink->dst;
-    ASyncContext       *s = ctx->priv;
-    AVFilterLink *outlink = ctx->outputs[0];
-    int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout);
-    int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
-                  av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
-    int out_size, ret;
-    int64_t delta;
-    int64_t new_pts;
-
-    /* buffer data until we get the next timestamp */
-    if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) {
-        if (pts != AV_NOPTS_VALUE) {
-            s->pts = pts - get_delay(s);
-        }
-        return write_to_fifo(s, buf);
-    }
-
-    if (s->first_pts != AV_NOPTS_VALUE) {
-        handle_trimming(ctx);
-        if (!avresample_available(s->avr))
-            return write_to_fifo(s, buf);
-    }
-
-    /* when we have two timestamps, compute how many samples would we have
-     * to add/remove to get proper sync between data and timestamps */
-    delta    = pts - s->pts - get_delay(s);
-    out_size = avresample_available(s->avr);
-
-    if (llabs(delta) > s->min_delta ||
-        (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) {
-        av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
-        out_size = av_clipl_int32((int64_t)out_size + delta);
-    } else {
-        if (s->resample) {
-            // adjust the compensation if delta is non-zero
-            int delay = get_delay(s);
-            int comp = s->comp + av_clip(delta * inlink->sample_rate / delay,
-                                         -s->max_comp, s->max_comp);
-            if (comp != s->comp) {
-                av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
-                if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) {
-                    s->comp = comp;
-                }
-            }
-        }
-        // adjust PTS to avoid monotonicity errors with input PTS jitter
-        pts -= delta;
-        delta = 0;
-    }
-
-    if (out_size > 0) {
-        AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size);
-        if (!buf_out) {
-            ret = AVERROR(ENOMEM);
-            goto fail;
-        }
-
-        if (s->first_frame && delta > 0) {
-            int planar = av_sample_fmt_is_planar(buf_out->format);
-            int planes = planar ?  nb_channels : 1;
-            int block_size = av_get_bytes_per_sample(buf_out->format) *
-                             (planar ? 1 : nb_channels);
-
-            int ch;
-
-            av_samples_set_silence(buf_out->extended_data, 0, delta,
-                                   nb_channels, buf->format);
-
-            for (ch = 0; ch < planes; ch++)
-                buf_out->extended_data[ch] += delta * block_size;
-
-            avresample_read(s->avr, buf_out->extended_data, out_size);
-
-            for (ch = 0; ch < planes; ch++)
-                buf_out->extended_data[ch] -= delta * block_size;
-        } else {
-            avresample_read(s->avr, buf_out->extended_data, out_size);
-
-            if (delta > 0) {
-                av_samples_set_silence(buf_out->extended_data, out_size - delta,
-                                       delta, nb_channels, buf->format);
-            }
-        }
-        buf_out->pts = s->pts;
-        ret = ff_filter_frame(outlink, buf_out);
-        if (ret < 0)
-            goto fail;
-        s->got_output = 1;
-    } else if (avresample_available(s->avr)) {
-        av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
-               "whole buffer.\n");
-    }
-
-    /* drain any remaining buffered data */
-    avresample_read(s->avr, NULL, avresample_available(s->avr));
-
-    new_pts = pts - avresample_get_delay(s->avr);
-    /* check for s->pts monotonicity */
-    if (new_pts > s->pts) {
-        s->pts = new_pts;
-        ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
-                                 buf->linesize[0], buf->nb_samples);
-    } else {
-        av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
-               "whole buffer.\n");
-        ret = 0;
-    }
-
-    s->first_frame = 0;
-fail:
-    av_frame_free(&buf);
-
-    return ret;
-}
-
-static const AVFilterPad avfilter_af_asyncts_inputs[] = {
-    {
-        .name          = "default",
-        .type          = AVMEDIA_TYPE_AUDIO,
-        .filter_frame  = filter_frame
-    },
-    { NULL }
-};
-
-static const AVFilterPad avfilter_af_asyncts_outputs[] = {
-    {
-        .name          = "default",
-        .type          = AVMEDIA_TYPE_AUDIO,
-        .config_props  = config_props,
-        .request_frame = request_frame
-    },
-    { NULL }
-};
-
-AVFilter ff_af_asyncts = {
-    .name        = "asyncts",
-    .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps."),
-    .init        = init,
-    .uninit      = uninit,
-    .priv_size   = sizeof(ASyncContext),
-    .priv_class  = &asyncts_class,
-    .query_formats = ff_query_formats_all_layouts,
-    .inputs      = avfilter_af_asyncts_inputs,
-    .outputs     = avfilter_af_asyncts_outputs,
-};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 93271fb2c4..64b634e8f3 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -79,7 +79,6 @@ static void register_all(void)
     REGISTER_FILTER(ASPLIT,         asplit,         af);
     REGISTER_FILTER(ASTATS,         astats,         af);
     REGISTER_FILTER(ASTREAMSELECT,  astreamselect,  af);
-    REGISTER_FILTER(ASYNCTS,        asyncts,        af);
     REGISTER_FILTER(ATEMPO,         atempo,         af);
     REGISTER_FILTER(ATRIM,          atrim,          af);
     REGISTER_FILTER(AZMQ,           azmq,           af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index a61ca32712..a798e911b0 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -31,7 +31,7 @@
 
 #define LIBAVFILTER_VERSION_MAJOR   6
 #define LIBAVFILTER_VERSION_MINOR  78
-#define LIBAVFILTER_VERSION_MICRO 100
+#define LIBAVFILTER_VERSION_MICRO 101
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \