avfilter: add loop filters

Signed-off-by: Paul B Mahol <onemda@gmail.com>

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:

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

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
 

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

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);

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 */

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, \

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;

libavutil/audio_fifo.h

@@ -110,6 +110,23 @@ 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.
  *