diff options
Diffstat (limited to 'libavfilter/audio.h')
| -rw-r--r-- | libavfilter/audio.h | 50 |
1 files changed, 46 insertions, 4 deletions
diff --git a/libavfilter/audio.h b/libavfilter/audio.h index 4684b6c..3335c96 100644 --- a/libavfilter/audio.h +++ b/libavfilter/audio.h @@ -1,18 +1,21 @@ /* - * This file is part of Libav. + * Copyright (c) Stefano Sabatini | stefasab at gmail.com + * Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -20,6 +23,25 @@ #define AVFILTER_AUDIO_H #include "avfilter.h" +#include "internal.h" + +static const enum AVSampleFormat ff_packed_sample_fmts_array[] = { + AV_SAMPLE_FMT_U8, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_S32, + AV_SAMPLE_FMT_FLT, + AV_SAMPLE_FMT_DBL, + AV_SAMPLE_FMT_NONE +}; + +static const enum AVSampleFormat ff_planar_sample_fmts_array[] = { + AV_SAMPLE_FMT_U8P, + AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE +}; /** default handler for get_audio_buffer() for audio inputs */ AVFrame *ff_default_get_audio_buffer(AVFilterLink *link, int nb_samples); @@ -38,4 +60,24 @@ AVFrame *ff_null_get_audio_buffer(AVFilterLink *link, int nb_samples); */ AVFrame *ff_get_audio_buffer(AVFilterLink *link, int nb_samples); +/** + * Send a buffer of audio samples to the next filter. + * + * @param link the output link over which the audio samples are being sent + * @param samplesref a reference to the buffer of audio samples being sent. The + * receiving filter will free this reference when it no longer + * needs it or pass it on to the next filter. + * + * @return >= 0 on success, a negative AVERROR on error. The receiving filter + * is responsible for unreferencing samplesref in case of error. + */ +int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); + +/** + * Send a buffer of audio samples to the next link, without checking + * min_samples. + */ +int ff_filter_samples_framed(AVFilterLink *link, + AVFilterBufferRef *samplesref); + #endif /* AVFILTER_AUDIO_H */ |
