summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
Diffstat
-rw-r--r--Changelog1
-rw-r--r--doc/APIchanges3
-rw-r--r--doc/filters.texi19
-rw-r--r--libavfilter/Makefile2
-rw-r--r--libavfilter/allfilters.c2
-rw-r--r--libavfilter/f_loop.c381
-rw-r--r--libavfilter/version.h2
-rw-r--r--libavutil/audio_fifo.c24
-rw-r--r--libavutil/audio_fifo.h17
9 files changed, 450 insertions, 1 deletions
diff --git a/Changelog b/Changelog
index 20dadcfe2c..b59db9225e 100644
--- a/Changelog
+++ b/Changelog
@@ -4,6 +4,7 @@ releases are sorted from youngest to oldest.
version <next>:
- DXVA2-accelerated HEVC Main10 decoding
- fieldhint filter
+- loop video filter and aloop audio filter
version 3.0:
diff --git a/doc/APIchanges b/doc/APIchanges
index fe6fff5d91..1194709fcd 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,6 +15,9 @@ libavutil: 2015-08-28
API changes, most recent first:
+2016-xx-xx - lavu 55.18.100
+ xxxxxxx audio_fifo.h - Add av_audio_fifo_peek_at().
+
2016-xx-xx - lavu 55.18.0
xxxxxxx buffer.h - Add av_buffer_pool_init2().
xxxxxxx hwcontext.h - Add a new installed header hwcontext.h with a new API
diff --git a/doc/filters.texi b/doc/filters.texi
index f30b9265b0..d5ff21cd5b 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -8185,6 +8185,25 @@ The formula that generates the correction is:
where @var{r_0} is halve of the image diagonal and @var{r_src} and @var{r_tgt} are the
distances from the focal point in the source and target images, respectively.
+@section loop, aloop
+
+Loop video frames or audio samples.
+
+Those filters accepts the following options:
+
+@table @option
+@item loop
+Set the number of loops.
+
+@item size
+Set maximal size in number of frames for @code{loop} filter or maximal number
+of samples in case of @code{aloop} filter.
+
+@item start
+Set first frame of loop for @code{loop} filter or first sample of loop in case
+of @code{aloop} filter.
+@end table
+
@anchor{lut3d}
@section lut3d
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 9120ecc70a..082ec49c5b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -38,6 +38,7 @@ OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
OBJS-$(CONFIG_ALIMITER_FILTER) += af_alimiter.o
OBJS-$(CONFIG_ALLPASS_FILTER) += af_biquads.o
+OBJS-$(CONFIG_ALOOP_FILTER) += f_loop.o
OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
OBJS-$(CONFIG_AMETADATA_FILTER) += f_metadata.o
OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
@@ -181,6 +182,7 @@ OBJS-$(CONFIG_INTERLACE_FILTER) += vf_interlace.o
OBJS-$(CONFIG_INTERLEAVE_FILTER) += f_interleave.o
OBJS-$(CONFIG_KERNDEINT_FILTER) += vf_kerndeint.o
OBJS-$(CONFIG_LENSCORRECTION_FILTER) += vf_lenscorrection.o
+OBJS-$(CONFIG_LOOP_FILTER) += f_loop.o
OBJS-$(CONFIG_LUT3D_FILTER) += vf_lut3d.o
OBJS-$(CONFIG_LUT_FILTER) += vf_lut.o
OBJS-$(CONFIG_LUTRGB_FILTER) += vf_lut.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 0fe72d6b44..4bce2afc4a 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -58,6 +58,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
REGISTER_FILTER(ALIMITER, alimiter, af);
REGISTER_FILTER(ALLPASS, allpass, af);
+ REGISTER_FILTER(ALOOP, aloop, af);
REGISTER_FILTER(AMERGE, amerge, af);
REGISTER_FILTER(AMETADATA, ametadata, af);
REGISTER_FILTER(AMIX, amix, af);
@@ -202,6 +203,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(INTERLEAVE, interleave, vf);
REGISTER_FILTER(KERNDEINT, kerndeint, vf);
REGISTER_FILTER(LENSCORRECTION, lenscorrection, vf);
+ REGISTER_FILTER(LOOP, loop, vf);
REGISTER_FILTER(LUT3D, lut3d, vf);
REGISTER_FILTER(LUT, lut, vf);
REGISTER_FILTER(LUTRGB, lutrgb, vf);
diff --git a/libavfilter/f_loop.c b/libavfilter/f_loop.c
new file mode 100644
index 0000000000..d8eb692e77
--- /dev/null
+++ b/libavfilter/f_loop.c
@@ -0,0 +1,381 @@
+/*
+ * Copyright (c) 2016 Paul B Mahol
+ *
+ * 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 "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/fifo.h"
+#include "libavutil/internal.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+#include "internal.h"
+#include "video.h"
+
+typedef struct LoopContext {
+ const AVClass *class;
+
+ AVAudioFifo *fifo;
+ AVAudioFifo *left;
+ AVFrame **frames;
+ int nb_frames;
+ int current_frame;
+ int64_t start_pts;
+ int64_t duration;
+ int64_t current_sample;
+ int64_t nb_samples;
+ int64_t ignored_samples;
+
+ int loop;
+ int64_t size;
+ int64_t start;
+ int64_t pts;
+} LoopContext;
+
+#define AFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define VFLAGS AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define OFFSET(x) offsetof(LoopContext, x)
+
+#if CONFIG_ALOOP_FILTER
+
+static int aconfig_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ LoopContext *s = ctx->priv;
+
+ s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, 8192);
+ s->left = av_audio_fifo_alloc(inlink->format, inlink->channels, 8192);
+ if (!s->fifo || !s->left)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static av_cold void auninit(AVFilterContext *ctx)
+{
+ LoopContext *s = ctx->priv;
+
+ av_audio_fifo_free(s->fifo);
+ av_audio_fifo_free(s->left);
+}
+
+static int push_samples(AVFilterContext *ctx, int nb_samples)
+{
+ AVFilterLink *outlink = ctx->outputs[0];
+ LoopContext *s = ctx->priv;
+ AVFrame *out;
+ int ret, i = 0;
+
+ while (s->loop != 0 && i < nb_samples) {
+ out = ff_get_audio_buffer(outlink, FFMIN(nb_samples, s->nb_samples - s->current_sample));
+ if (!out)
+ return AVERROR(ENOMEM);
+ ret = av_audio_fifo_peek_at(s->fifo, (void **)out->extended_data, out->nb_samples, s->current_sample);
+ if (ret < 0)
+ return ret;
+ out->pts = s->pts;
+ out->nb_samples = ret;
+ s->pts += out->nb_samples;
+ i += out->nb_samples;
+ s->current_sample += out->nb_samples;
+
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ return ret;
+
+ if (s->current_sample >= s->nb_samples) {
+ s->current_sample = 0;
+
+ if (s->loop > 0)
+ s->loop--;
+ }
+ }
+
+ return ret;
+}
+
+static int afilter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ LoopContext *s = ctx->priv;
+ int ret = 0;
+
+ if (s->ignored_samples + frame->nb_samples > s->start && s->size > 0 && s->loop != 0) {
+ if (s->nb_samples < s->size) {
+ int written = FFMIN(frame->nb_samples, s->size - s->nb_samples);
+ int drain = 0;
+
+ ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, written);
+ if (ret < 0)
+ return ret;
+ if (!s->nb_samples) {
+ drain = FFMAX(0, s->start - s->ignored_samples);
+ s->pts = frame->pts;
+ av_audio_fifo_drain(s->fifo, drain);
+ s->pts += s->start - s->ignored_samples;
+ }
+ s->nb_samples += ret - drain;
+ drain = frame->nb_samples - written;
+ if (s->nb_samples == s->size && drain > 0) {
+ int ret2;
+
+ ret2 = av_audio_fifo_write(s->left, (void **)frame->extended_data, frame->nb_samples);
+ if (ret2 < 0)
+ return ret2;
+ av_audio_fifo_drain(s->left, drain);
+ }
+ frame->nb_samples = ret;
+ s->pts += ret;
+ ret = ff_filter_frame(outlink, frame);
+ } else {
+ int nb_samples = frame->nb_samples;
+
+ av_frame_free(&frame);
+ ret = push_samples(ctx, nb_samples);
+ }
+ } else {
+ s->ignored_samples += frame->nb_samples;
+ frame->pts = s->pts;
+ s->pts += frame->nb_samples;
+ ret = ff_filter_frame(outlink, frame);
+ }
+
+ return ret;
+}
+
+static int arequest_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ LoopContext *s = ctx->priv;
+ int ret = 0;
+
+ if ((!s->size) ||
+ (s->nb_samples < s->size) ||
+ (s->nb_samples >= s->size && s->loop == 0)) {
+ int nb_samples = av_audio_fifo_size(s->left);
+
+ if (s->loop == 0 && nb_samples > 0) {
+ AVFrame *out;
+
+ out = ff_get_audio_buffer(outlink, nb_samples);
+ if (!out)
+ return AVERROR(ENOMEM);
+ av_audio_fifo_read(s->left, (void **)out->extended_data, nb_samples);
+ out->pts = s->pts;
+ s->pts += nb_samples;
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ return ret;
+ }
+ ret = ff_request_frame(ctx->inputs[0]);
+ } else {
+ ret = push_samples(ctx, 1024);
+ }
+
+ if (ret == AVERROR_EOF && s->nb_samples > 0 && s->loop != 0) {
+ ret = push_samples(ctx, outlink->sample_rate);
+ }
+
+ return ret;
+}
+
+static const AVOption aloop_options[] = {
+ { "loop", "number of loops", OFFSET(loop), AV_OPT_TYPE_INT, {.i64 = 0 }, -1, INT_MAX, AFLAGS },
+ { "size", "max number of samples to loop", OFFSET(size), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT32_MAX, AFLAGS },
+ { "start", "set the loop start sample", OFFSET(start), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, AFLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aloop);
+
+static const AVFilterPad ainputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = afilter_frame,
+ .config_props = aconfig_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad aoutputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = arequest_frame,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_aloop = {
+ .name = "aloop",
+ .description = NULL_IF_CONFIG_SMALL("Loop audio samples."),
+ .priv_size = sizeof(LoopContext),
+ .priv_class = &aloop_class,
+ .uninit = auninit,
+ .query_formats = ff_query_formats_all,
+ .inputs = ainputs,
+ .outputs = aoutputs,
+};
+#endif /* CONFIG_ALOOP_FILTER */
+
+#if CONFIG_LOOP_FILTER
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ LoopContext *s = ctx->priv;
+
+ s->frames = av_calloc(s->size, sizeof(*s->frames));
+ if (!s->frames)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ LoopContext *s = ctx->priv;
+ int i;
+
+ for (i = 0; i < s->nb_frames; i++)
+ av_frame_free(&s->frames[i]);
+
+ av_freep(&s->frames);
+ s->nb_frames = 0;
+}
+
+static int push_frame(AVFilterContext *ctx)
+{
+ AVFilterLink *outlink = ctx->outputs[0];
+ LoopContext *s = ctx->priv;
+ int64_t pts;
+ int ret;
+
+ AVFrame *out = av_frame_clone(s->frames[s->current_frame]);
+
+ if (!out)
+ return AVERROR(ENOMEM);
+ out->pts += s->duration - s->start_pts;
+ pts = out->pts + av_frame_get_pkt_duration(out);
+ ret = ff_filter_frame(outlink, out);
+ s->current_frame++;
+
+ if (s->current_frame >= s->nb_frames) {
+ s->duration = pts;
+ s->current_frame = 0;
+
+ if (s->loop > 0)
+ s->loop--;
+ }
+
+ return ret;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ LoopContext *s = ctx->priv;
+ int ret = 0;
+
+ if (inlink->frame_count >= s->start && s->size > 0 && s->loop != 0) {
+ if (s->nb_frames < s->size) {
+ if (!s->nb_frames)
+ s->start_pts = frame->pts;
+ s->frames[s->nb_frames] = av_frame_clone(frame);
+ if (!s->frames[s->nb_frames]) {
+ av_frame_free(&frame);
+ return AVERROR(ENOMEM);
+ }
+ s->nb_frames++;
+ s->duration = frame->pts + av_frame_get_pkt_duration(frame);
+ ret = ff_filter_frame(outlink, frame);
+ } else {
+ av_frame_free(&frame);
+ ret = push_frame(ctx);
+ }
+ } else {
+ frame->pts += s->duration;
+ ret = ff_filter_frame(outlink, frame);
+ }
+
+ return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ LoopContext *s = ctx->priv;
+ int ret = 0;
+
+ if ((!s->size) ||
+ (s->nb_frames < s->size) ||
+ (s->nb_frames >= s->size && s->loop == 0)) {
+ ret = ff_request_frame(ctx->inputs[0]);
+ } else {
+ ret = push_frame(ctx);
+ }
+
+ if (ret == AVERROR_EOF && s->nb_frames > 0 && s->loop != 0) {
+ ret = push_frame(ctx);
+ }
+
+ return ret;
+}
+
+static const AVOption loop_options[] = {
+ { "loop", "number of loops", OFFSET(loop), AV_OPT_TYPE_INT, {.i64 = 0 }, -1, INT_MAX, VFLAGS },
+ { "size", "max number of frames to loop", OFFSET(size), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT16_MAX, VFLAGS },
+ { "start", "set the loop start frame", OFFSET(start), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, VFLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(loop);
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_VIDEO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_VIDEO,
+ .request_frame = request_frame,
+ },
+ { NULL }
+};
+
+AVFilter ff_vf_loop = {
+ .name = "loop",
+ .description = NULL_IF_CONFIG_SMALL("Loop video frames."),
+ .priv_size = sizeof(LoopContext),
+ .priv_class = &loop_class,
+ .init = init,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+};
+#endif /* CONFIG_LOOP_FILTER */
diff --git a/libavfilter/version.h b/libavfilter/version.h
index fe0539c145..7dc1033a42 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
-#define LIBAVFILTER_VERSION_MINOR 32
+#define LIBAVFILTER_VERSION_MINOR 33
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
diff --git a/libavutil/audio_fifo.c b/libavutil/audio_fifo.c
index d5298cce4d..e4d38e0524 100644
--- a/libavutil/audio_fifo.c
+++ b/libavutil/audio_fifo.c
@@ -155,6 +155,30 @@ int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
return nb_samples;
}
+int av_audio_fifo_peek_at(AVAudioFifo *af, void **data, int nb_samples, int offset)
+{
+ int i, ret, size;
+
+ if (offset < 0 || offset >= af->nb_samples)
+ return AVERROR(EINVAL);
+ if (nb_samples < 0)
+ return AVERROR(EINVAL);
+ nb_samples = FFMIN(nb_samples, af->nb_samples);
+ if (!nb_samples)
+ return 0;
+ if (offset > af->nb_samples - nb_samples)
+ return AVERROR(EINVAL);
+
+ offset *= af->sample_size;
+ size = nb_samples * af->sample_size;
+ for (i = 0; i < af->nb_buffers; i++) {
+ if ((ret = av_fifo_generic_peek_at(af->buf[i], data[i], offset, size, NULL)) < 0)
+ return AVERROR_BUG;
+ }
+
+ return nb_samples;
+}
+
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
{
int i, ret, size;
diff --git a/libavutil/audio_fifo.h b/libavutil/audio_fifo.h
index 24f91dab72..d8a9194a8d 100644
--- a/libavutil/audio_fifo.h
+++ b/libavutil/audio_fifo.h
@@ -111,6 +111,23 @@ int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples);
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples);
/**
+ * Peek data from an AVAudioFifo.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param af AVAudioFifo to read from
+ * @param data audio data plane pointers
+ * @param nb_samples number of samples to peek
+ * @param offset offset from current read position
+ * @return number of samples actually peek, or negative AVERROR code
+ * on failure. The number of samples actually peek will not
+ * be greater than nb_samples, and will only be less than
+ * nb_samples if av_audio_fifo_size is less than nb_samples.
+ */
+int av_audio_fifo_peek_at(AVAudioFifo *af, void **data, int nb_samples, int offset);
+
+/**
* Read data from an AVAudioFifo.
*
* @see enum AVSampleFormat
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-16 14:28:22 +0000