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