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 : - 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 @@ -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. *