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FFmpeg/libavfilter/audio.c
Andreas Rheinhardt 50ea7389ec avfilter: Deduplicate default audio inputs/outputs
Lots of audio filters use very simple inputs or outputs:
An array with a single AVFilterPad whose name is "default"
and whose type is AVMEDIA_TYPE_AUDIO; everything else is unset.

Given that we never use pointer equality for inputs or outputs*,
we can simply use a single AVFilterPad instead of dozens; this
even saves .data.rel.ro (4784B here) as well as relocations.

*: In fact, several filters (like the filters in af_biquads.c)
already use the same inputs; furthermore, ff_filter_alloc()
duplicates the input and output pads so that we do not even
work with the pads directly.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-08-07 09:21:13 +02:00

119 lines
3.8 KiB
C

/*
* Copyright (c) Stefano Sabatini | stefasab at gmail.com
* Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu
*
* 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
*/
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/cpu.h"
#include "audio.h"
#include "avfilter.h"
#include "framepool.h"
#include "internal.h"
const AVFilterPad ff_audio_default_filterpad[1] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
}
};
AVFrame *ff_null_get_audio_buffer(AVFilterLink *link, int nb_samples)
{
return ff_get_audio_buffer(link->dst->outputs[0], nb_samples);
}
AVFrame *ff_default_get_audio_buffer(AVFilterLink *link, int nb_samples)
{
AVFrame *frame = NULL;
int channels = link->ch_layout.nb_channels;
int align = av_cpu_max_align();
#if FF_API_OLD_CHANNEL_LAYOUT
FF_DISABLE_DEPRECATION_WARNINGS
int channel_layout_nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
av_assert0(channels == channel_layout_nb_channels || !channel_layout_nb_channels);
FF_ENABLE_DEPRECATION_WARNINGS
#endif
if (!link->frame_pool) {
link->frame_pool = ff_frame_pool_audio_init(av_buffer_allocz, channels,
nb_samples, link->format, align);
if (!link->frame_pool)
return NULL;
} else {
int pool_channels = 0;
int pool_nb_samples = 0;
int pool_align = 0;
enum AVSampleFormat pool_format = AV_SAMPLE_FMT_NONE;
if (ff_frame_pool_get_audio_config(link->frame_pool,
&pool_channels, &pool_nb_samples,
&pool_format, &pool_align) < 0) {
return NULL;
}
if (pool_channels != channels || pool_nb_samples < nb_samples ||
pool_format != link->format || pool_align != align) {
ff_frame_pool_uninit((FFFramePool **)&link->frame_pool);
link->frame_pool = ff_frame_pool_audio_init(av_buffer_allocz, channels,
nb_samples, link->format, align);
if (!link->frame_pool)
return NULL;
}
}
frame = ff_frame_pool_get(link->frame_pool);
if (!frame)
return NULL;
frame->nb_samples = nb_samples;
#if FF_API_OLD_CHANNEL_LAYOUT
FF_DISABLE_DEPRECATION_WARNINGS
frame->channel_layout = link->channel_layout;
FF_ENABLE_DEPRECATION_WARNINGS
#endif
if (link->ch_layout.order != AV_CHANNEL_ORDER_UNSPEC &&
av_channel_layout_copy(&frame->ch_layout, &link->ch_layout) < 0) {
av_frame_free(&frame);
return NULL;
}
frame->sample_rate = link->sample_rate;
av_samples_set_silence(frame->extended_data, 0, nb_samples, channels, link->format);
return frame;
}
AVFrame *ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
{
AVFrame *ret = NULL;
if (link->dstpad->get_buffer.audio)
ret = link->dstpad->get_buffer.audio(link, nb_samples);
if (!ret)
ret = ff_default_get_audio_buffer(link, nb_samples);
return ret;
}