From: Stefano Sabatini <[email protected]> This filter changes the number of samples on single output operation.
Based on a patch by Andrey Utkin <[email protected]>. Signed-off-by: Alex Converse <[email protected]> --- doc/filters.texi | 30 ++++++ libavfilter/Makefile | 1 + libavfilter/af_asetnsamples.c | 201 +++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 233 insertions(+), 0 deletions(-) create mode 100644 libavfilter/af_asetnsamples.c diff --git a/doc/filters.texi b/doc/filters.texi index e77256e..364d664 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -175,6 +175,36 @@ stream ends. The default value is 2 seconds. Pass the audio source unchanged to the output. +@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 asplit Split input audio into several identical outputs. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 530aa57..7be84ba 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -28,6 +28,7 @@ OBJS-$(CONFIG_AFIFO_FILTER) += fifo.o OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o +OBJS-$(CONFIG_ASETNSAMPLES_FILTER) += af_asetnsamples.o OBJS-$(CONFIG_ASPLIT_FILTER) += split.o OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o diff --git a/libavfilter/af_asetnsamples.c b/libavfilter/af_asetnsamples.c new file mode 100644 index 0000000..6157886 --- /dev/null +++ b/libavfilter/af_asetnsamples.c @@ -0,0 +1,201 @@ +/* + * Copyright (c) 2012 Andrey Utkin + * Copyright (c) 2012 Stefano Sabatini + * + * This file is part of Libav. + * + * 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/audioconvert.h" +#include "libavutil/avassert.h" +#include "libavutil/common.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "internal.h" +#include "formats.h" + +typedef struct { + 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) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM + +static const AVOption asetnsamples_options[] = { +{ "pad", "pad last frame with silence", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1, FLAGS }, +{ "p", "pad last frame with silence", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1, FLAGS }, +{ "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX, FLAGS }, +{ "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX, FLAGS }, +{ NULL } +}; + +static av_cold int asetnsamples_init(AVFilterContext *ctx, const char *args) +{ + ASNSContext *asns = ctx->priv; + int err; + + av_opt_set_defaults(asns); + + if ((err = av_set_options_string(asns, args, "=", ":")) < 0) + return err; + + asns->next_out_pts = AV_NOPTS_VALUE; + av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad); + + return 0; +} + +static av_cold void asetnsamples_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 int 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 -1; + } + } + 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); + + while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples) + push_samples(outlink); + return 0; +} + +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 = ff_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 = asetnsamples_init, + .uninit = asetnsamples_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 94b3115..53cb05f 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -39,6 +39,7 @@ void avfilter_register_all(void) REGISTER_FILTER (AFORMAT, aformat, af); REGISTER_FILTER (AMIX, amix, af); REGISTER_FILTER (ANULL, anull, af); + REGISTER_FILTER (ASETNSAMPLES,asetnsamples,af); REGISTER_FILTER (ASPLIT, asplit, af); REGISTER_FILTER (ASYNCTS, asyncts, af); REGISTER_FILTER (CHANNELMAP, channelmap, af); -- 1.7.7.3 _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
