On 2/11/16, Paul B Mahol <one...@gmail.com> wrote:
> Hi,
>
> patch attached.
>
Better version attached.
From d953f78bffbf3db8f0209b41b189ece12b402afa Mon Sep 17 00:00:00 2001
From: Paul B Mahol <one...@gmail.com>
Date: Thu, 11 Feb 2016 22:05:54 +0100
Subject: [PATCH] avfilter: add loop filters
Signed-off-by: Paul B Mahol <one...@gmail.com>
---
libavfilter/Makefile | 2 +
libavfilter/allfilters.c | 2 +
libavfilter/f_loop.c | 339 +++++++++++++++++++++++++++++++++++++++++++++++
libavutil/audio_fifo.c | 24 ++++
libavutil/audio_fifo.h | 17 +++
5 files changed, 384 insertions(+)
create mode 100644 libavfilter/f_loop.c
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8916588..35ac53a 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
@@ -180,6 +181,7 @@ 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 fa7d304..6331fe5 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);
@@ -201,6 +202,7 @@ 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 0000000..b902af0
--- /dev/null
+++ b/libavfilter/f_loop.c
@@ -0,0 +1,339 @@
+/*
+ * 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;
+ 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);
+ if (!s->fifo)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static av_cold void auninit(AVFilterContext *ctx)
+{
+ LoopContext *s = ctx->priv;
+
+ av_audio_fifo_free(s->fifo);
+}
+
+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;
+ 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;
+ ret = ff_filter_frame(outlink, frame);
+ }
+
+ return ret;
+}
+
+static int arequest_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ LoopContext *s = ctx->priv;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ 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 {
+ ret = ff_filter_frame(outlink, frame);
+ }
+
+ return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ LoopContext *s = ctx->priv;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ 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/libavutil/audio_fifo.c b/libavutil/audio_fifo.c
index d5298cc..e4d38e0 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 24f91da..d8a9194 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
--
1.9.1
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
