lavfi: remove af_asynts filter

Long overdue for removal, af_aresample should be used instead.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>

Changelog

@@ -2,6 +2,7 @@ Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
 version <next>:
+- Removed asyncts filter (use af_aresample instead)
 - CrystalHD decoder moved to new decode API
 - add internal ebur128 library, remove external libebur128 dependency
 - Pro-MPEG CoP #3-R2 FEC protocol

configure

@@ -3076,7 +3076,6 @@ afftfilt_filter_select="fft"
 amovie_filter_deps="avcodec avformat"
 aresample_filter_deps="swresample"
 ass_filter_deps="libass"
-asyncts_filter_deps="avresample"
 atempo_filter_deps="avcodec"
 atempo_filter_select="rdft"
 azmq_filter_deps="libzmq"
@@ -6459,7 +6458,6 @@ enabled zlib && add_cppflags -DZLIB_CONST
 enabled afftfilt_filter     && prepend avfilter_deps "avcodec"
 enabled amovie_filter       && prepend avfilter_deps "avformat avcodec"
 enabled aresample_filter    && prepend avfilter_deps "swresample"
-enabled asyncts_filter      && prepend avfilter_deps "avresample"
 enabled atempo_filter       && prepend avfilter_deps "avcodec"
 enabled cover_rect_filter   && prepend avfilter_deps "avformat avcodec"
 enabled ebur128_filter && enabled swresample && prepend avfilter_deps "swresample"

doc/filters.texi

@@ -1642,39 +1642,6 @@ Number of occasions (not the number of samples) that the signal attained either
 Overall bit depth of audio. Number of bits used for each sample.
 @end table
 
-@section asyncts
-
-Synchronize audio data with timestamps by squeezing/stretching it and/or
-dropping samples/adding silence when needed.
-
-This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
-
-It accepts the following parameters:
-@table @option
-
-@item compensate
-Enable stretching/squeezing the data to make it match the timestamps. Disabled
-by default. When disabled, time gaps are covered with silence.
-
-@item min_delta
-The minimum difference between timestamps and audio data (in seconds) to trigger
-adding/dropping samples. The default value is 0.1. If you get an imperfect
-sync with this filter, try setting this parameter to 0.
-
-@item max_comp
-The maximum compensation in samples per second. Only relevant with compensate=1.
-The default value is 500.
-
-@item first_pts
-Assume that the first PTS should be this value. The time base is 1 / sample
-rate. This allows for padding/trimming at the start of the stream. By default,
-no assumption is made about the first frame's expected PTS, so no padding or
-trimming is done. For example, this could be set to 0 to pad the beginning with
-silence if an audio stream starts after the video stream or to trim any samples
-with a negative PTS due to encoder delay.
-
-@end table
-
 @section atempo
 
 Adjust audio tempo.

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

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);

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, \

tests/fate/filter-audio.mak

@@ -187,12 +187,6 @@ $(FATE_AMIX): SRC1 = $(TARGET_PATH)/tests/data/asynth-44100-2-2.wav
 $(FATE_AMIX): CMP  = oneoff
 $(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-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