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325f6e0a97
* qatar/master: lavfi: do not export the filters from shared objects Conflicts: libavfilter/af_amix.c libavfilter/af_anull.c libavfilter/asrc_anullsrc.c libavfilter/f_select.c libavfilter/f_settb.c libavfilter/split.c libavfilter/src_movie.c libavfilter/vf_aspect.c libavfilter/vf_blackframe.c libavfilter/vf_colorbalance.c libavfilter/vf_copy.c libavfilter/vf_crop.c libavfilter/vf_cropdetect.c libavfilter/vf_drawbox.c libavfilter/vf_format.c libavfilter/vf_framestep.c libavfilter/vf_frei0r.c libavfilter/vf_hflip.c libavfilter/vf_libopencv.c libavfilter/vf_lut.c libavfilter/vf_null.c libavfilter/vf_overlay.c libavfilter/vf_scale.c libavfilter/vf_transpose.c libavfilter/vf_unsharp.c libavfilter/vf_vflip.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
322 lines
10 KiB
C
322 lines
10 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavresample/avresample.h"
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#include "libavutil/attributes.h"
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#include "libavutil/audio_fifo.h"
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#include "libavutil/common.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct ASyncContext {
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const AVClass *class;
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AVAudioResampleContext *avr;
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int64_t pts; ///< timestamp in samples of the first sample in fifo
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int min_delta; ///< pad/trim min threshold in samples
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int first_frame; ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE
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int64_t first_pts; ///< user-specified first expected pts, in samples
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int comp; ///< current resample compensation
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/* options */
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int resample;
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float min_delta_sec;
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int max_comp;
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/* set by filter_frame() to signal an output frame to request_frame() */
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int got_output;
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} ASyncContext;
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#define OFFSET(x) offsetof(ASyncContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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#define F AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption asyncts_options[] = {
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{ "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, A|F },
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{ "min_delta", "Minimum difference between timestamps and audio data "
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"(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A|F },
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{ "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A|F },
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{ "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 },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(asyncts);
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static av_cold int init(AVFilterContext *ctx)
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{
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ASyncContext *s = ctx->priv;
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s->pts = AV_NOPTS_VALUE;
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s->first_frame = 1;
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ASyncContext *s = ctx->priv;
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if (s->avr) {
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avresample_close(s->avr);
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avresample_free(&s->avr);
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}
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}
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static int config_props(AVFilterLink *link)
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{
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ASyncContext *s = link->src->priv;
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int ret;
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s->min_delta = s->min_delta_sec * link->sample_rate;
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link->time_base = (AVRational){1, link->sample_rate};
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s->avr = avresample_alloc_context();
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if (!s->avr)
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return AVERROR(ENOMEM);
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av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0);
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av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0);
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av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0);
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av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0);
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av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0);
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av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0);
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if (s->resample)
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av_opt_set_int(s->avr, "force_resampling", 1, 0);
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if ((ret = avresample_open(s->avr)) < 0)
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return ret;
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return 0;
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}
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/* get amount of data currently buffered, in samples */
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static int64_t get_delay(ASyncContext *s)
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{
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return avresample_available(s->avr) + avresample_get_delay(s->avr);
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}
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static void handle_trimming(AVFilterContext *ctx)
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{
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ASyncContext *s = ctx->priv;
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if (s->pts < s->first_pts) {
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int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr));
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av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n",
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delta);
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avresample_read(s->avr, NULL, delta);
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s->pts += delta;
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} else if (s->first_frame)
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s->pts = s->first_pts;
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}
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static int request_frame(AVFilterLink *link)
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{
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AVFilterContext *ctx = link->src;
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ASyncContext *s = ctx->priv;
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int ret = 0;
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int nb_samples;
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s->got_output = 0;
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while (ret >= 0 && !s->got_output)
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ret = ff_request_frame(ctx->inputs[0]);
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/* flush the fifo */
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if (ret == AVERROR_EOF) {
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if (s->first_pts != AV_NOPTS_VALUE)
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handle_trimming(ctx);
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if (nb_samples = get_delay(s)) {
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AVFrame *buf = ff_get_audio_buffer(link, nb_samples);
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if (!buf)
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return AVERROR(ENOMEM);
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ret = avresample_convert(s->avr, buf->extended_data,
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buf->linesize[0], nb_samples, NULL, 0, 0);
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if (ret <= 0) {
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av_frame_free(&buf);
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return (ret < 0) ? ret : AVERROR_EOF;
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}
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buf->pts = s->pts;
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return ff_filter_frame(link, buf);
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}
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}
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return ret;
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}
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static int write_to_fifo(ASyncContext *s, AVFrame *buf)
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{
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int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
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buf->linesize[0], buf->nb_samples);
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av_frame_free(&buf);
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return ret;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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ASyncContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout);
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int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
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av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
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int out_size, ret;
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int64_t delta;
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int64_t new_pts;
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/* buffer data until we get the next timestamp */
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if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) {
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if (pts != AV_NOPTS_VALUE) {
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s->pts = pts - get_delay(s);
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}
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return write_to_fifo(s, buf);
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}
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if (s->first_pts != AV_NOPTS_VALUE) {
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handle_trimming(ctx);
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if (!avresample_available(s->avr))
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return write_to_fifo(s, buf);
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}
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/* when we have two timestamps, compute how many samples would we have
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* to add/remove to get proper sync between data and timestamps */
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delta = pts - s->pts - get_delay(s);
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out_size = avresample_available(s->avr);
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if (labs(delta) > s->min_delta ||
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(s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) {
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av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
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out_size = av_clipl_int32((int64_t)out_size + delta);
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} else {
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if (s->resample) {
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// adjust the compensation if delta is non-zero
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int delay = get_delay(s);
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int comp = s->comp + av_clip(delta * inlink->sample_rate / delay,
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-s->max_comp, s->max_comp);
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if (comp != s->comp) {
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av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
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if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) {
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s->comp = comp;
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}
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}
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}
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// adjust PTS to avoid monotonicity errors with input PTS jitter
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pts -= delta;
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delta = 0;
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}
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if (out_size > 0) {
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AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size);
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if (!buf_out) {
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ret = AVERROR(ENOMEM);
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goto fail;
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}
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if (s->first_frame && delta > 0) {
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int planar = av_sample_fmt_is_planar(buf_out->format);
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int planes = planar ? nb_channels : 1;
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int block_size = av_get_bytes_per_sample(buf_out->format) *
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(planar ? 1 : nb_channels);
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int ch;
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av_samples_set_silence(buf_out->extended_data, 0, delta,
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nb_channels, buf->format);
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for (ch = 0; ch < planes; ch++)
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buf_out->extended_data[ch] += delta * block_size;
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avresample_read(s->avr, buf_out->extended_data, out_size);
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for (ch = 0; ch < planes; ch++)
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buf_out->extended_data[ch] -= delta * block_size;
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} else {
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avresample_read(s->avr, buf_out->extended_data, out_size);
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if (delta > 0) {
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av_samples_set_silence(buf_out->extended_data, out_size - delta,
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delta, nb_channels, buf->format);
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}
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}
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buf_out->pts = s->pts;
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ret = ff_filter_frame(outlink, buf_out);
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if (ret < 0)
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goto fail;
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s->got_output = 1;
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} else if (avresample_available(s->avr)) {
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av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
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"whole buffer.\n");
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}
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/* drain any remaining buffered data */
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avresample_read(s->avr, NULL, avresample_available(s->avr));
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new_pts = pts - avresample_get_delay(s->avr);
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/* check for s->pts monotonicity */
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if (new_pts > s->pts) {
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s->pts = new_pts;
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ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
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buf->linesize[0], buf->nb_samples);
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} else {
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av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
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"whole buffer.\n");
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ret = 0;
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}
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s->first_frame = 0;
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fail:
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av_frame_free(&buf);
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return ret;
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}
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static const AVFilterPad avfilter_af_asyncts_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame
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},
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{ NULL }
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};
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static const AVFilterPad avfilter_af_asyncts_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_props,
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.request_frame = request_frame
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},
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{ NULL }
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};
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AVFilter ff_af_asyncts = {
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.name = "asyncts",
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.description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),
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.init = init,
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.uninit = uninit,
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.priv_size = sizeof(ASyncContext),
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.priv_class = &asyncts_class,
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.inputs = avfilter_af_asyncts_inputs,
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.outputs = avfilter_af_asyncts_outputs,
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};
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