diff options
Diffstat (limited to 'libavfilter/audio.h')
| -rw-r--r-- | libavfilter/audio.h | 36 |
1 files changed, 32 insertions, 4 deletions
diff --git a/libavfilter/audio.h b/libavfilter/audio.h index fa448e2fd6..a84c378ec8 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 */ @@ -21,6 +24,24 @@ #include "avfilter.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 */ AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms, int nb_samples); @@ -55,4 +76,11 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms, */ 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 */ |