/* * 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/audioconvert.h" #include "libavutil/avassert.h" #include "libavutil/opt.h" #include "avfilter.h" #include "audio.h" #include "internal.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 asetnsamples_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 } }; AVFILTER_DEFINE_CLASS(asetnsamples); static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque) { ASNSContext *asns = ctx->priv; int err; asns->class = &asetnsamples_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 = 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 = 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 } }, };