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avfilter: add loop filters 

Signed-off-by: Paul B Mahol <onemda@gmail.com>
tags/n3.1
Paul B Mahol 9 years ago
parent
5590ab45e0
commit
08acab85d3
9 changed files with 450 additions and 1 deletions
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  1. +1
    -0
      Changelog
  2. +3
    -0
      doc/APIchanges
  3. +19
    -0
      doc/filters.texi
  4. +2
    -0
      libavfilter/Makefile
  5. +2
    -0
      libavfilter/allfilters.c
  6. +381
    -0
      libavfilter/f_loop.c
  7. +1
    -1
      libavfilter/version.h
  8. +24
    -0
      libavutil/audio_fifo.c
  9. +17
    -0
      libavutil/audio_fifo.h

+ 1
- 0
Changelog View File

@@ -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:


+ 3
- 0
doc/APIchanges View File

@@ -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


+ 19
- 0
doc/filters.texi View File

@@ -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



+ 2
- 0
libavfilter/Makefile View File

@@ -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


+ 2
- 0
libavfilter/allfilters.c View File

@@ -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);


+ 381
- 0
libavfilter/f_loop.c View File

@@ -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 */

+ 1
- 1
libavfilter/version.h View File

@@ -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, \


+ 24
- 0
libavutil/audio_fifo.c View File

@@ -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;


+ 17
- 0
libavutil/audio_fifo.h View File

@@ -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.
*


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