mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-27 11:51:49 +02:00
208 lines
6.6 KiB
C
208 lines
6.6 KiB
C
/*
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* Copyright (c) 2012 Andrey Utkin
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* Copyright (c) 2012 Stefano Sabatini
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Filter that changes number of samples on single output operation
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*/
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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#include "formats.h"
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typedef struct {
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const AVClass *class;
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int nb_out_samples; ///< how many samples to output
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AVAudioFifo *fifo; ///< samples are queued here
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int64_t next_out_pts;
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int req_fullfilled;
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int pad;
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} ASNSContext;
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#define OFFSET(x) offsetof(ASNSContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption asetnsamples_options[] = {
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{ "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
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{ "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
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{ "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
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{ "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(asetnsamples);
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static av_cold int init(AVFilterContext *ctx, const char *args)
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{
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ASNSContext *asns = ctx->priv;
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int err;
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asns->class = &asetnsamples_class;
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av_opt_set_defaults(asns);
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if ((err = av_set_options_string(asns, args, "=", ":")) < 0)
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return err;
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asns->next_out_pts = AV_NOPTS_VALUE;
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av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ASNSContext *asns = ctx->priv;
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av_audio_fifo_free(asns->fifo);
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}
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static int config_props_output(AVFilterLink *outlink)
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{
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ASNSContext *asns = outlink->src->priv;
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int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
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asns->fifo = av_audio_fifo_alloc(outlink->format, nb_channels, asns->nb_out_samples);
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if (!asns->fifo)
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return AVERROR(ENOMEM);
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return 0;
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}
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static int push_samples(AVFilterLink *outlink)
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{
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ASNSContext *asns = outlink->src->priv;
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AVFilterBufferRef *outsamples = NULL;
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int nb_out_samples, nb_pad_samples;
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if (asns->pad) {
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nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
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nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
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} else {
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nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
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nb_pad_samples = 0;
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}
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if (!nb_out_samples)
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return 0;
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outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_out_samples);
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av_assert0(outsamples);
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av_audio_fifo_read(asns->fifo,
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(void **)outsamples->extended_data, nb_out_samples);
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if (nb_pad_samples)
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av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
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nb_pad_samples, av_get_channel_layout_nb_channels(outlink->channel_layout),
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outlink->format);
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outsamples->audio->nb_samples = nb_out_samples;
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outsamples->audio->channel_layout = outlink->channel_layout;
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outsamples->audio->sample_rate = outlink->sample_rate;
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outsamples->pts = asns->next_out_pts;
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if (asns->next_out_pts != AV_NOPTS_VALUE)
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asns->next_out_pts += nb_out_samples;
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ff_filter_frame(outlink, outsamples);
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asns->req_fullfilled = 1;
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return nb_out_samples;
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}
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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AVFilterContext *ctx = inlink->dst;
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ASNSContext *asns = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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int ret;
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int nb_samples = insamples->audio->nb_samples;
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if (av_audio_fifo_space(asns->fifo) < nb_samples) {
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av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
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ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
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if (ret < 0) {
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av_log(ctx, AV_LOG_ERROR,
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"Stretching audio fifo failed, discarded %d samples\n", nb_samples);
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return -1;
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}
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}
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av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
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if (asns->next_out_pts == AV_NOPTS_VALUE)
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asns->next_out_pts = insamples->pts;
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avfilter_unref_buffer(insamples);
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while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
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push_samples(outlink);
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return 0;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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ASNSContext *asns = outlink->src->priv;
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AVFilterLink *inlink = outlink->src->inputs[0];
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int ret;
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asns->req_fullfilled = 0;
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do {
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ret = ff_request_frame(inlink);
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} while (!asns->req_fullfilled && ret >= 0);
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if (ret == AVERROR_EOF)
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while (push_samples(outlink))
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;
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return ret;
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}
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static const AVFilterPad asetnsamples_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ | AV_PERM_WRITE,
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},
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{ NULL }
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};
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static const AVFilterPad asetnsamples_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.request_frame = request_frame,
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.config_props = config_props_output,
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},
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{ NULL }
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};
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AVFilter avfilter_af_asetnsamples = {
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.name = "asetnsamples",
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.description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
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.priv_size = sizeof(ASNSContext),
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.init = init,
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.uninit = uninit,
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.inputs = asetnsamples_inputs,
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.outputs = asetnsamples_outputs,
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.priv_class = &asetnsamples_class,
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};
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