/* * Copyright (c) 2012 Michael Niedermayer * * 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 * audio pad filter. * * Based on af_aresample.c */ #include "libavutil/avstring.h"#include "libavutil/channel_layout.h"#include "libavutil/opt.h"#include "libavutil/samplefmt.h"#include "libavutil/avassert.h"#include "avfilter.h"#include "audio.h"#include "filters.h" typedef struct APadContext { const AVClass *class; int64_t next_pts; int eof; int packet_size; int64_t pad_len, pad_len_left; int64_t whole_len, whole_len_left; int64_t pad_dur; int64_t whole_dur;} APadContext; #define OFFSET(x) offsetof(APadContext, x)#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption apad_options[] = { { "packet_size", "set silence packet size", OFFSET(packet_size), AV_OPT_TYPE_INT, { .i64 = 4096 }, 0, INT_MAX, A }, { "pad_len", "set number of samples of silence to add", OFFSET(pad_len), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A }, { "whole_len", "set minimum target number of samples in the audio stream", OFFSET(whole_len), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A }, { "pad_dur", "set duration of silence to add", OFFSET(pad_dur), AV_OPT_TYPE_DURATION, { .i64 = -1 }, -1, INT64_MAX, A }, { "whole_dur", "set minimum target duration in the audio stream", OFFSET(whole_dur), AV_OPT_TYPE_DURATION, { .i64 = -1 }, -1, INT64_MAX, A }, { NULL }}; AVFILTER_DEFINE_CLASS(apad); static av_cold int init(AVFilterContext *ctx){ APadContext *s = ctx->priv; s->next_pts = AV_NOPTS_VALUE; if (s->whole_len >= 0 && s->pad_len >= 0) { av_log(ctx, AV_LOG_ERROR, "Both whole and pad length are set, this is not possible\n"); return AVERROR(EINVAL); } return 0;} static int filter_frame(AVFilterLink *inlink, AVFrame *frame){ AVFilterContext *ctx = inlink->dst; APadContext *s = ctx->priv; if (s->whole_len >= 0) { s->whole_len_left = FFMAX(s->whole_len_left - frame->nb_samples, 0); av_log(ctx, AV_LOG_DEBUG, "n_out:%d whole_len_left:%"PRId64"\n", frame->nb_samples, s->whole_len_left); } s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base); return ff_filter_frame(ctx->outputs[0], frame);} static int push_frame(AVFilterLink *outlink){ AVFilterContext *ctx = outlink->src; APadContext *s = ctx->priv; AVFrame *outsamplesref; int n_out; if (ctx->is_disabled) return 0; n_out = s->packet_size; if (s->whole_len >= 0 && s->pad_len < 0) { s->pad_len = s->pad_len_left = s->whole_len_left; } if (s->pad_len >=0 || s->whole_len >= 0) { n_out = FFMIN(n_out, s->pad_len_left); s->pad_len_left -= n_out; av_log(ctx, AV_LOG_DEBUG, "padding n_out:%d pad_len_left:%"PRId64"\n", n_out, s->pad_len_left); } if (!n_out) return AVERROR_EOF; outsamplesref = ff_get_audio_buffer(outlink, n_out); if (!outsamplesref) return AVERROR(ENOMEM); av_assert0(outsamplesref->sample_rate == outlink->sample_rate); av_assert0(outsamplesref->nb_samples == n_out); av_samples_set_silence(outsamplesref->extended_data, 0, n_out, outsamplesref->ch_layout.nb_channels, outsamplesref->format); outsamplesref->pts = s->next_pts; if (s->next_pts != AV_NOPTS_VALUE) s->next_pts += av_rescale_q(n_out, (AVRational){1, outlink->sample_rate}, outlink->time_base); return ff_filter_frame(outlink, outsamplesref);} static int activate(AVFilterContext *ctx){ AVFilterLink *inlink = ctx->inputs[0]; AVFilterLink *outlink = ctx->outputs[0]; APadContext *s = ctx->priv; int64_t pts; int status; FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); if (!s->eof && ff_inlink_queued_frames(inlink)) { AVFrame *frame = NULL; int ret; ret = ff_inlink_consume_frame(inlink, &frame); if (ret < 0) return ret; if (ret > 0) return filter_frame(inlink, frame); } if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) s->eof = status == AVERROR_EOF; if (s->eof) { int ret = push_frame(outlink); if (ret == AVERROR_EOF) { ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts); return 0; } return ret; } FF_FILTER_FORWARD_WANTED(outlink, inlink); return FFERROR_NOT_READY;} static int config_output(AVFilterLink *outlink){ AVFilterContext *ctx = outlink->src; APadContext *s = ctx->priv; if (s->pad_dur >= 0) s->pad_len = av_rescale(s->pad_dur, outlink->sample_rate, AV_TIME_BASE); if (s->whole_dur >= 0) s->whole_len = av_rescale(s->whole_dur, outlink->sample_rate, AV_TIME_BASE); s->pad_len_left = s->pad_len; s->whole_len_left = s->whole_len; return 0;} static const AVFilterPad apad_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, },}; const FFFilter ff_af_apad = { .p.name = "apad", .p.description = NULL_IF_CONFIG_SMALL("Pad audio with silence."), .p.priv_class = &apad_class, .p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, .init = init, .activate = activate, .priv_size = sizeof(APadContext), FILTER_INPUTS(ff_audio_default_filterpad), FILTER_OUTPUTS(apad_outputs),};
/*
* Copyright (c) 2012 Michael Niedermayer
*
* 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
* audio pad filter.
*
* Based on af_aresample.c
*/
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
typedef struct APadContext {
const AVClass *class;
int64_t next_pts;
int eof;
int packet_size;
int64_t pad_len, pad_len_left;
int64_t whole_len, whole_len_left;
int64_t pad_dur;
int64_t whole_dur;
} APadContext;
#define OFFSET(x) offsetof(APadContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption apad_options[] = {
{ "packet_size", "set silence packet size", OFFSET(packet_size), AV_OPT_TYPE_INT, { .i64 = 4096 }, 0, INT_MAX, A },
{ "pad_len", "set number of samples of silence to add", OFFSET(pad_len), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A },
{ "whole_len", "set minimum target number of samples in the audio stream", OFFSET(whole_len), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A },
{ "pad_dur", "set duration of silence to add", OFFSET(pad_dur), AV_OPT_TYPE_DURATION, { .i64 = -1 }, -1, INT64_MAX, A },
{ "whole_dur", "set minimum target duration in the audio stream", OFFSET(whole_dur), AV_OPT_TYPE_DURATION, { .i64 = -1 }, -1, INT64_MAX, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(apad);
static av_cold int init(AVFilterContext *ctx)
{
APadContext *s = ctx->priv;
s->next_pts = AV_NOPTS_VALUE;
if (s->whole_len >= 0 && s->pad_len >= 0) {
av_log(ctx, AV_LOG_ERROR, "Both whole and pad length are set, this is not possible\n");
return AVERROR(EINVAL);
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
APadContext *s = ctx->priv;
if (s->whole_len >= 0) {
s->whole_len_left = FFMAX(s->whole_len_left - frame->nb_samples, 0);
av_log(ctx, AV_LOG_DEBUG,
"n_out:%d whole_len_left:%"PRId64"\n", frame->nb_samples, s->whole_len_left);
}
s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
return ff_filter_frame(ctx->outputs[0], frame);
}
static int push_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
APadContext *s = ctx->priv;
AVFrame *outsamplesref;
int n_out;
if (ctx->is_disabled)
return 0;
n_out = s->packet_size;
if (s->whole_len >= 0 && s->pad_len < 0) {
s->pad_len = s->pad_len_left = s->whole_len_left;
}
if (s->pad_len >=0 || s->whole_len >= 0) {
n_out = FFMIN(n_out, s->pad_len_left);
s->pad_len_left -= n_out;
av_log(ctx, AV_LOG_DEBUG,
"padding n_out:%d pad_len_left:%"PRId64"\n", n_out, s->pad_len_left);
}
if (!n_out)
return AVERROR_EOF;
outsamplesref = ff_get_audio_buffer(outlink, n_out);
if (!outsamplesref)
return AVERROR(ENOMEM);
av_assert0(outsamplesref->sample_rate == outlink->sample_rate);
av_assert0(outsamplesref->nb_samples == n_out);
av_samples_set_silence(outsamplesref->extended_data, 0,
n_out,
outsamplesref->ch_layout.nb_channels,
outsamplesref->format);
outsamplesref->pts = s->next_pts;
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += av_rescale_q(n_out, (AVRational){1, outlink->sample_rate}, outlink->time_base);
return ff_filter_frame(outlink, outsamplesref);
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
APadContext *s = ctx->priv;
int64_t pts;
int status;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (!s->eof && ff_inlink_queued_frames(inlink)) {
AVFrame *frame = NULL;
int ret;
ret = ff_inlink_consume_frame(inlink, &frame);
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, frame);
}
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts))
s->eof = status == AVERROR_EOF;
if (s->eof) {
int ret = push_frame(outlink);
if (ret == AVERROR_EOF) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
return 0;
}
return ret;
}
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
APadContext *s = ctx->priv;
if (s->pad_dur >= 0)
s->pad_len = av_rescale(s->pad_dur, outlink->sample_rate, AV_TIME_BASE);
if (s->whole_dur >= 0)
s->whole_len = av_rescale(s->whole_dur, outlink->sample_rate, AV_TIME_BASE);
s->pad_len_left = s->pad_len;
s->whole_len_left = s->whole_len;
return 0;
}
static const AVFilterPad apad_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const FFFilter ff_af_apad = {
.p.name = "apad",
.p.description = NULL_IF_CONFIG_SMALL("Pad audio with silence."),
.p.priv_class = &apad_class,
.p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
.init = init,
.activate = activate,
.priv_size = sizeof(APadContext),
FILTER_INPUTS(ff_audio_default_filterpad),
FILTER_OUTPUTS(apad_outputs),
};