From 9f26421b0be2af36b5405608f4e7429b4bd7fbdb Mon Sep 17 00:00:00 2001 From: Anton Khirnov Date: Tue, 8 May 2012 16:33:50 +0200 Subject: [PATCH] lavfi: add asyncts filter. --- doc/filters.texi | 19 ++++ libavfilter/Makefile | 2 + libavfilter/af_asyncts.c | 237 +++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 259 insertions(+) create mode 100644 libavfilter/af_asyncts.c diff --git a/doc/filters.texi b/doc/filters.texi index f066657add..0e611d2793 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -137,6 +137,25 @@ aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo Pass the audio source unchanged to the output. +@section asyncts +Synchronize audio data with timestamps by squeezing/stretching it and/or +dropping samples/adding silence when needed. + +The filter accepts the following named parameters: +@table @option + +@item compensate +Enable stretching/squeezing the data to make it match the timestamps. + +@item min_delta +Minimum difference between timestamps and audio data (in seconds) to trigger +adding/dropping samples. + +@item max_comp +Maximum compensation in samples per second. + +@end table + @section resample Convert the audio sample format, sample rate and channel layout. This filter is not meant to be used directly, it is inserted automatically by libavfilter diff --git a/libavfilter/Makefile b/libavfilter/Makefile index df75bd5e74..a90d8a02b1 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -1,5 +1,6 @@ NAME = avfilter FFLIBS = avutil swscale +FFLIBS-$(CONFIG_ASYNCTS_FILTER) += avresample FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample @@ -24,6 +25,7 @@ OBJS = allfilters.o \ OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o +OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o diff --git a/libavfilter/af_asyncts.c b/libavfilter/af_asyncts.c new file mode 100644 index 0000000000..5cde0bf00a --- /dev/null +++ b/libavfilter/af_asyncts.c @@ -0,0 +1,237 @@ +/* + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavresample/avresample.h" +#include "libavutil/audio_fifo.h" +#include "libavutil/mathematics.h" +#include "libavutil/opt.h" +#include "libavutil/samplefmt.h" + +#include "audio.h" +#include "avfilter.h" + +typedef struct ASyncContext { + const AVClass *class; + + AVAudioResampleContext *avr; + int64_t pts; ///< timestamp in samples of the first sample in fifo + int min_delta; ///< pad/trim min threshold in samples + + /* options */ + int resample; + float min_delta_sec; + int max_comp; +} ASyncContext; + +#define OFFSET(x) offsetof(ASyncContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM +static const AVOption options[] = { + { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { 0 }, 0, 1, A }, + { "min_delta", "Minimum difference between timestamps and audio data " + "(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { 0.1 }, 0, INT_MAX, A }, + { "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { 500 }, 0, INT_MAX, A }, + { NULL }, +}; + +static const AVClass async_class = { + .class_name = "asyncts filter", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +static int init(AVFilterContext *ctx, const char *args, void *opaque) +{ + ASyncContext *s = ctx->priv; + int ret; + + s->class = &async_class; + av_opt_set_defaults(s); + + if ((ret = av_set_options_string(s, args, "=", ":")) < 0) { + av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args); + return ret; + } + av_opt_free(s); + + s->pts = AV_NOPTS_VALUE; + + return 0; +} + +static void uninit(AVFilterContext *ctx) +{ + ASyncContext *s = ctx->priv; + + if (s->avr) { + avresample_close(s->avr); + avresample_free(&s->avr); + } +} + +static int config_props(AVFilterLink *link) +{ + ASyncContext *s = link->src->priv; + int ret; + + s->min_delta = s->min_delta_sec * link->sample_rate; + link->time_base = (AVRational){1, link->sample_rate}; + + s->avr = avresample_alloc_context(); + if (!s->avr) + return AVERROR(ENOMEM); + + av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0); + av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0); + av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0); + av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0); + av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0); + av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0); + + if (s->resample) + av_opt_set_int(s->avr, "force_resampling", 1, 0); + + if ((ret = avresample_open(s->avr)) < 0) + return ret; + + return 0; +} + +static int request_frame(AVFilterLink *link) +{ + AVFilterContext *ctx = link->src; + ASyncContext *s = ctx->priv; + int ret = avfilter_request_frame(ctx->inputs[0]); + int nb_samples; + + /* flush the fifo */ + if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) { + AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE, + nb_samples); + if (!buf) + return AVERROR(ENOMEM); + avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0], + nb_samples, NULL, 0, 0); + buf->pts = s->pts; + ff_filter_samples(link, buf); + return 0; + } + + return ret; +} + +static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf) +{ + avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, + buf->linesize[0], buf->audio->nb_samples); + avfilter_unref_buffer(buf); +} + +/* get amount of data currently buffered, in samples */ +static int64_t get_delay(ASyncContext *s) +{ + return avresample_available(s->avr) + avresample_get_delay(s->avr); +} + +static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) +{ + AVFilterContext *ctx = inlink->dst; + ASyncContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout); + int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts : + av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); + int out_size; + int64_t delta; + + /* buffer data until we get the first timestamp */ + if (s->pts == AV_NOPTS_VALUE) { + if (pts != AV_NOPTS_VALUE) { + s->pts = pts - get_delay(s); + } + write_to_fifo(s, buf); + return; + } + + /* now wait for the next timestamp */ + if (pts == AV_NOPTS_VALUE) { + write_to_fifo(s, buf); + return; + } + + /* when we have two timestamps, compute how many samples would we have + * to add/remove to get proper sync between data and timestamps */ + delta = pts - s->pts - get_delay(s); + out_size = avresample_available(s->avr); + + if (labs(delta) > s->min_delta) { + av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta); + out_size += delta; + } else if (s->resample) { + int comp = av_clip(delta, -s->max_comp, s->max_comp); + av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp); + avresample_set_compensation(s->avr, delta, inlink->sample_rate); + } + + if (out_size > 0) { + AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, + out_size); + if (!buf_out) + return; + + avresample_read(s->avr, (void**)buf_out->extended_data, out_size); + buf_out->pts = s->pts; + + if (delta > 0) { + av_samples_set_silence(buf_out->extended_data, out_size - delta, + delta, nb_channels, buf->format); + } + ff_filter_samples(outlink, buf_out); + } else { + av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " + "whole buffer.\n"); + } + + /* drain any remaining buffered data */ + avresample_read(s->avr, NULL, avresample_available(s->avr)); + + s->pts = pts - avresample_get_delay(s->avr); + avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, + buf->linesize[0], buf->audio->nb_samples); + avfilter_unref_buffer(buf); +} + +AVFilter avfilter_af_asyncts = { + .name = "asyncts", + .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"), + + .init = init, + .uninit = uninit, + + .priv_size = sizeof(ASyncContext), + + .inputs = (const AVFilterPad[]) {{ .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_samples = filter_samples }, + { NULL }}, + .outputs = (const AVFilterPad[]) {{ .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_props, + .request_frame = request_frame }, + { NULL }}, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 4f5f852b8b..3fa0152d86 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -36,6 +36,7 @@ void avfilter_register_all(void) REGISTER_FILTER (AFORMAT, aformat, af); REGISTER_FILTER (ANULL, anull, af); + REGISTER_FILTER (ASYNCTS, asyncts, af); REGISTER_FILTER (RESAMPLE, resample, af); REGISTER_FILTER (ANULLSRC, anullsrc, asrc);