@@ -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
@@ -3074,7 +3074,6 @@ 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"
@@ -6460,7 +6459,6 @@ 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"
@@ -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.
@@ -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
deleted file mode 100644
@@ -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,
-};
@@ -84,7 +84,6 @@ void avfilter_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);
@@ -31,7 +31,7 @@
#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 75
-#define LIBAVFILTER_VERSION_MICRO 100
+#define LIBAVFILTER_VERSION_MICRO 101
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
@@ -187,12 +187,6 @@ $(FATE_AMIX): SRC1 = $(TARGET_PATH)/tests/data/asynth-44100-2-2.wav
$(FATE_AMIX): CMP = oneoff
$(FATE_AMIX): CMP_UNIT = f32
-FATE_AFILTER_SAMPLES-$(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