diff --git a/Changelog b/Changelog index 026d10290b104f015b6241c2f448649a07f30cc7..facd0e9b13a5f9cccb9e7149530825b780dbf0bc 100644 --- a/Changelog +++ b/Changelog @@ -6,6 +6,7 @@ version next: - Scene detection in libavfilter - Indeo Audio decoder - channelsplit audio filter +- setnsamples audio filter version 0.11: diff --git a/doc/filters.texi b/doc/filters.texi index 67242af1364a4e3b7dd4ce5c84f5c19361395fe0..f2767cb9743048f7951d64c0d90e3de77f32823b 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -273,6 +273,36 @@ For example, to resample the input audio to 44100Hz: aresample=44100 @end example +@section asetnsamples + +Set the number of samples per each output audio frame. + +The last output packet may contain a different number of samples, as +the filter will flush all the remaining samples when the input audio +signal its end. + +The filter accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +@table @option + +@item nb_out_samples, n +Set the number of frames per each output audio frame. The number is +intended as the number of samples @emph{per each channel}. +Default value is 1024. + +@item pad, p +If set to 1, the filter will pad the last audio frame with zeroes, so +that the last frame will contain the same number of samples as the +previous ones. Default value is 1. +@end table + +For example, to set the number of per-frame samples to 1234 and +disable padding for the last frame, use: +@example +asetnsamples=n=1234:p=0 +@end example + @section ashowinfo Show a line containing various information for each input audio frame. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 95126a8b1d15a1394fc03eb1605c6e0bce5f493f..fee7641f5076f40514b10f86905a5dcd6056044c 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -51,6 +51,7 @@ OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o +OBJS-$(CONFIG_ASETNSAMPLES_FILTER) += af_asetnsamples.o OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o OBJS-$(CONFIG_ASPLIT_FILTER) += split.o OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o diff --git a/libavfilter/af_asetnsamples.c b/libavfilter/af_asetnsamples.c new file mode 100644 index 0000000000000000000000000000000000000000..eb3c6a9360354ee28ae23097f55d2447b965f2c0 --- /dev/null +++ b/libavfilter/af_asetnsamples.c @@ -0,0 +1,206 @@ +/* + * Copyright (c) 2012 Andrey Utkin + * Copyright (c) 2012 Stefano Sabatini + * + * 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 + */ + +/** + * @file + * Filter that changes number of samples on single output operation + */ + +#include "libavutil/audio_fifo.h" +#include "libavutil/avassert.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +typedef struct { + const AVClass *class; + int nb_out_samples; ///< how many samples to output + AVAudioFifo *fifo; ///< samples are queued here + int64_t next_out_pts; + int req_fullfilled; + int pad; +} ASNSContext; + +#define OFFSET(x) offsetof(ASNSContext, x) + +static const AVOption asns_options[] = { +{ "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 }, +{ "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 }, +{ "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX }, +{ "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX }, +{ NULL } +}; + +static const AVClass asns_class = { + "asetnsamples", + av_default_item_name, + asns_options +}; + +static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque) +{ + ASNSContext *asns = ctx->priv; + int err; + + asns->class = &asns_class; + av_opt_set_defaults(asns); + + if ((err = av_set_options_string(asns, args, "=", ":")) < 0) { + av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args); + return err; + } + + asns->next_out_pts = AV_NOPTS_VALUE; + av_log(ctx, AV_LOG_INFO, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad); + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + ASNSContext *asns = ctx->priv; + av_audio_fifo_free(asns->fifo); +} + +static int config_props_output(AVFilterLink *outlink) +{ + ASNSContext *asns = outlink->src->priv; + int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout); + + asns->fifo = av_audio_fifo_alloc(outlink->format, nb_channels, asns->nb_out_samples); + if (!asns->fifo) + return AVERROR(ENOMEM); + + return 0; +} + +static int push_samples(AVFilterLink *outlink) +{ + ASNSContext *asns = outlink->src->priv; + AVFilterBufferRef *outsamples = NULL; + int nb_out_samples, nb_pad_samples; + + if (asns->pad) { + nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0; + nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo)); + } else { + nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo)); + nb_pad_samples = 0; + } + + if (!nb_out_samples) + return 0; + + outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_out_samples); + av_assert0(outsamples); + + av_audio_fifo_read(asns->fifo, + (void **)outsamples->extended_data, nb_out_samples); + + if (nb_pad_samples) + av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples, + nb_pad_samples, av_get_channel_layout_nb_channels(outlink->channel_layout), + outlink->format); + outsamples->audio->nb_samples = nb_out_samples; + outsamples->audio->channel_layout = outlink->channel_layout; + outsamples->audio->sample_rate = outlink->sample_rate; + outsamples->pts = asns->next_out_pts; + + if (asns->next_out_pts != AV_NOPTS_VALUE) + asns->next_out_pts += nb_out_samples; + + ff_filter_samples(outlink, outsamples); + asns->req_fullfilled = 1; + return nb_out_samples; +} + +static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +{ + AVFilterContext *ctx = inlink->dst; + ASNSContext *asns = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + int ret; + int nb_samples = insamples->audio->nb_samples; + + if (av_audio_fifo_space(asns->fifo) < nb_samples) { + av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples); + ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples); + if (ret < 0) { + av_log(ctx, AV_LOG_ERROR, + "Stretching audio fifo failed, discarded %d samples\n", nb_samples); + return; + } + } + av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples); + if (asns->next_out_pts == AV_NOPTS_VALUE) + asns->next_out_pts = insamples->pts; + avfilter_unref_buffer(insamples); + + if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples) + push_samples(outlink); +} + +static int request_frame(AVFilterLink *outlink) +{ + ASNSContext *asns = outlink->src->priv; + AVFilterLink *inlink = outlink->src->inputs[0]; + int ret; + + asns->req_fullfilled = 0; + do { + ret = avfilter_request_frame(inlink); + } while (!asns->req_fullfilled && ret >= 0); + + if (ret == AVERROR_EOF) + while (push_samples(outlink)) + ; + + return ret; +} + +AVFilter avfilter_af_asetnsamples = { + .name = "asetnsamples", + .description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."), + .priv_size = sizeof(ASNSContext), + .init = init, + .uninit = uninit, + + .inputs = (const AVFilterPad[]) { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_samples = filter_samples, + .min_perms = AV_PERM_READ|AV_PERM_WRITE + }, + { .name = NULL } + }, + + .outputs = (const AVFilterPad[]) { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .request_frame = request_frame, + .config_props = config_props_output, + }, + { .name = NULL } + }, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index f8d6b389b43b598847d9cb944761d8172823f993..8257ee540f6f7328406e7988ec97d2686aa222d2 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -40,6 +40,7 @@ void avfilter_register_all(void) REGISTER_FILTER (AMIX, amix, af); REGISTER_FILTER (ANULL, anull, af); REGISTER_FILTER (ARESAMPLE, aresample, af); + REGISTER_FILTER (ASETNSAMPLES, asetnsamples, af); REGISTER_FILTER (ASHOWINFO, ashowinfo, af); REGISTER_FILTER (ASPLIT, asplit, af); REGISTER_FILTER (ASTREAMSYNC, astreamsync, af); diff --git a/libavfilter/version.h b/libavfilter/version.h index 3ebea25e07081c9e5faa4ab0b2b976272d533ec4..861fe99a0deb358c75e5213edbea95c59f95a4cd 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -29,7 +29,7 @@ #include "libavutil/avutil.h" #define LIBAVFILTER_VERSION_MAJOR 2 -#define LIBAVFILTER_VERSION_MINOR 79 +#define LIBAVFILTER_VERSION_MINOR 80 #define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \