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https://gitee.com/openharmony/third_party_ffmpeg
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lavfi: remove af_asynts filter
Long overdue for removal, af_aresample should be used instead. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit is contained in:
Rostislav Pehlivanov
Rostislav Pehlivanov
parent
d7896e9b42
commit
a8fe8d6b4a
@ -2,6 +2,7 @@ Entries are sorted chronologically from oldest to youngest within each release,
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releases are sorted from youngest to oldest.
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releases are sorted from youngest to oldest.
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version <next>:
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version <next>:
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- Removed asyncts filter (use af_aresample instead)
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- CrystalHD decoder moved to new decode API
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- CrystalHD decoder moved to new decode API
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- add internal ebur128 library, remove external libebur128 dependency
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- add internal ebur128 library, remove external libebur128 dependency
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- Pro-MPEG CoP #3-R2 FEC protocol
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- Pro-MPEG CoP #3-R2 FEC protocol
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2
configure
vendored
2
configure
vendored
@ -3076,7 +3076,6 @@ afftfilt_filter_select="fft"
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amovie_filter_deps="avcodec avformat"
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amovie_filter_deps="avcodec avformat"
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aresample_filter_deps="swresample"
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aresample_filter_deps="swresample"
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ass_filter_deps="libass"
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ass_filter_deps="libass"
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asyncts_filter_deps="avresample"
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atempo_filter_deps="avcodec"
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atempo_filter_deps="avcodec"
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atempo_filter_select="rdft"
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atempo_filter_select="rdft"
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azmq_filter_deps="libzmq"
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azmq_filter_deps="libzmq"
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@ -6459,7 +6458,6 @@ enabled zlib && add_cppflags -DZLIB_CONST
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enabled afftfilt_filter && prepend avfilter_deps "avcodec"
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enabled afftfilt_filter && prepend avfilter_deps "avcodec"
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enabled amovie_filter && prepend avfilter_deps "avformat avcodec"
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enabled amovie_filter && prepend avfilter_deps "avformat avcodec"
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enabled aresample_filter && prepend avfilter_deps "swresample"
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enabled aresample_filter && prepend avfilter_deps "swresample"
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enabled asyncts_filter && prepend avfilter_deps "avresample"
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enabled atempo_filter && prepend avfilter_deps "avcodec"
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enabled atempo_filter && prepend avfilter_deps "avcodec"
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enabled cover_rect_filter && prepend avfilter_deps "avformat avcodec"
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enabled cover_rect_filter && prepend avfilter_deps "avformat avcodec"
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enabled ebur128_filter && enabled swresample && prepend avfilter_deps "swresample"
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enabled ebur128_filter && enabled swresample && prepend avfilter_deps "swresample"
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@ -1642,39 +1642,6 @@ Number of occasions (not the number of samples) that the signal attained either
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Overall bit depth of audio. Number of bits used for each sample.
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Overall bit depth of audio. Number of bits used for each sample.
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@end table
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@end table
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@section asyncts
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Synchronize audio data with timestamps by squeezing/stretching it and/or
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dropping samples/adding silence when needed.
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This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
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It accepts the following parameters:
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@table @option
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@item compensate
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Enable stretching/squeezing the data to make it match the timestamps. Disabled
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by default. When disabled, time gaps are covered with silence.
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@item min_delta
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The minimum difference between timestamps and audio data (in seconds) to trigger
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adding/dropping samples. The default value is 0.1. If you get an imperfect
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sync with this filter, try setting this parameter to 0.
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@item max_comp
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The maximum compensation in samples per second. Only relevant with compensate=1.
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The default value is 500.
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@item first_pts
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Assume that the first PTS should be this value. The time base is 1 / sample
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rate. This allows for padding/trimming at the start of the stream. By default,
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no assumption is made about the first frame's expected PTS, so no padding or
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trimming is done. For example, this could be set to 0 to pad the beginning with
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silence if an audio stream starts after the video stream or to trim any samples
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with a negative PTS due to encoder delay.
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@end table
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@section atempo
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@section atempo
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Adjust audio tempo.
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Adjust audio tempo.
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@ -67,7 +67,6 @@ OBJS-$(CONFIG_ASIDEDATA_FILTER) += f_sidedata.o
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OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
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OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
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OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
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OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
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OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o
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OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o
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OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o
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OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
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OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
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OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
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OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
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OBJS-$(CONFIG_AZMQ_FILTER) += f_zmq.o
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OBJS-$(CONFIG_AZMQ_FILTER) += f_zmq.o
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@ -1,323 +0,0 @@
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/*
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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 <stdint.h>
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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_BOOL, { .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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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 (llabs(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) {
|
|
||||||
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,
|
|
||||||
};
|
|
||||||
@ -79,7 +79,6 @@ static void register_all(void)
|
|||||||
REGISTER_FILTER(ASPLIT, asplit, af);
|
REGISTER_FILTER(ASPLIT, asplit, af);
|
||||||
REGISTER_FILTER(ASTATS, astats, af);
|
REGISTER_FILTER(ASTATS, astats, af);
|
||||||
REGISTER_FILTER(ASTREAMSELECT, astreamselect, af);
|
REGISTER_FILTER(ASTREAMSELECT, astreamselect, af);
|
||||||
REGISTER_FILTER(ASYNCTS, asyncts, af);
|
|
||||||
REGISTER_FILTER(ATEMPO, atempo, af);
|
REGISTER_FILTER(ATEMPO, atempo, af);
|
||||||
REGISTER_FILTER(ATRIM, atrim, af);
|
REGISTER_FILTER(ATRIM, atrim, af);
|
||||||
REGISTER_FILTER(AZMQ, azmq, af);
|
REGISTER_FILTER(AZMQ, azmq, af);
|
||||||
|
|||||||
@ -31,7 +31,7 @@
|
|||||||
|
|
||||||
#define LIBAVFILTER_VERSION_MAJOR 6
|
#define LIBAVFILTER_VERSION_MAJOR 6
|
||||||
#define LIBAVFILTER_VERSION_MINOR 78
|
#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, \
|
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
|
||||||
LIBAVFILTER_VERSION_MINOR, \
|
LIBAVFILTER_VERSION_MINOR, \
|
||||||
|
|||||||
@ -187,12 +187,6 @@ $(FATE_AMIX): SRC1 = $(TARGET_PATH)/tests/data/asynth-44100-2-2.wav
|
|||||||
$(FATE_AMIX): CMP = oneoff
|
$(FATE_AMIX): CMP = oneoff
|
||||||
$(FATE_AMIX): CMP_UNIT = f32
|
$(FATE_AMIX): CMP_UNIT = f32
|
||||||
|
|
||||||
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECMUX, ASYNCTS, FLV, NELLYMOSER, PCM_S16LE) += fate-filter-asyncts
|
|
||||||
fate-filter-asyncts: SRC = $(TARGET_SAMPLES)/nellymoser/nellymoser-discont.flv
|
|
||||||
fate-filter-asyncts: CMD = pcm -analyzeduration 10000000 -i $(SRC) -af asyncts
|
|
||||||
fate-filter-asyncts: CMP = oneoff
|
|
||||||
fate-filter-asyncts: REF = $(SAMPLES)/nellymoser/nellymoser-discont-async-v3.pcm
|
|
||||||
|
|
||||||
FATE_AFILTER_SAMPLES-$(CONFIG_ARESAMPLE_FILTER) += fate-filter-aresample
|
FATE_AFILTER_SAMPLES-$(CONFIG_ARESAMPLE_FILTER) += fate-filter-aresample
|
||||||
fate-filter-aresample: SRC = $(TARGET_SAMPLES)/nellymoser/nellymoser-discont.flv
|
fate-filter-aresample: SRC = $(TARGET_SAMPLES)/nellymoser/nellymoser-discont.flv
|
||||||
fate-filter-aresample: CMD = pcm -analyzeduration 10000000 -i $(SRC) -af aresample=min_comp=0.001:min_hard_comp=0.1:first_pts=0
|
fate-filter-aresample: CMD = pcm -analyzeduration 10000000 -i $(SRC) -af aresample=min_comp=0.001:min_hard_comp=0.1:first_pts=0
|
||||||
|
|||||||
Loading…
Reference in New Issue
Block a user