/* * 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 "audio.h" #include "avfilter.h" #include "internal.h" 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 = av_frame_alloc(); int channels = link->channels; int ret; av_assert0(channels == av_get_channel_layout_nb_channels(link->channel_layout) || !av_get_channel_layout_nb_channels(link->channel_layout)); if (!frame) return NULL; frame->nb_samples = nb_samples; frame->format = link->format; av_frame_set_channels(frame, link->channels); frame->channel_layout = link->channel_layout; frame->sample_rate = link->sample_rate; ret = av_frame_get_buffer(frame, 0); if (ret < 0) { av_frame_free(&frame); return NULL; } 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_audio_buffer) ret = link->dstpad->get_audio_buffer(link, nb_samples); if (!ret) ret = ff_default_get_audio_buffer(link, nb_samples); return ret; }