ffmpeg/libavfilter/audio.c
Anton Khirnov 6d75d44d90 lavfi: drop internal.h
All that remains in it are things that belong in avfilter_internal.h.

Move them there and remove internal.h
2024-08-19 21:48:04 +02:00

145 lines
4.4 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 "libavutil/eval.h"
#include "audio.h"
#include "avfilter.h"
#include "avfilter_internal.h"
#include "filters.h"
#include "framepool.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;
FilterLinkInternal *const li = ff_link_internal(link);
int channels = link->ch_layout.nb_channels;
int align = av_cpu_max_align();
if (!li->frame_pool) {
li->frame_pool = ff_frame_pool_audio_init(av_buffer_allocz, channels,
nb_samples, link->format, align);
if (!li->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(li->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(&li->frame_pool);
li->frame_pool = ff_frame_pool_audio_init(av_buffer_allocz, channels,
nb_samples, link->format, align);
if (!li->frame_pool)
return NULL;
}
}
frame = ff_frame_pool_get(li->frame_pool);
if (!frame)
return NULL;
frame->nb_samples = nb_samples;
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;
}
int ff_parse_sample_rate(int *ret, const char *arg, void *log_ctx)
{
char *tail;
double srate = av_strtod(arg, &tail);
if (*tail || srate < 1 || (int)srate != srate || srate > INT_MAX) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid sample rate '%s'\n", arg);
return AVERROR(EINVAL);
}
*ret = srate;
return 0;
}
int ff_parse_channel_layout(AVChannelLayout *ret, int *nret, const char *arg,
void *log_ctx)
{
AVChannelLayout chlayout = { 0 };
int res;
res = av_channel_layout_from_string(&chlayout, arg);
if (res < 0) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid channel layout '%s'\n", arg);
return AVERROR(EINVAL);
}
if (chlayout.order == AV_CHANNEL_ORDER_UNSPEC && !nret) {
av_log(log_ctx, AV_LOG_ERROR, "Unknown channel layout '%s' is not supported.\n", arg);
return AVERROR(EINVAL);
}
*ret = chlayout;
if (nret)
*nret = chlayout.nb_channels;
return 0;
}