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https://github.com/xenia-project/FFmpeg.git
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587c8ab912
Originally based on code by Stefano Sabatini and S. N. Hemanth. Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
81 lines
3.0 KiB
C
81 lines
3.0 KiB
C
/*
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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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#ifndef AVFILTER_ASRC_ABUFFER_H
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#define AVFILTER_ASRC_ABUFFER_H
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#include "avfilter.h"
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/**
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* @file
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* memory buffer source for audio
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*/
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/**
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* Queue an audio buffer to the audio buffer source.
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*
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* @param abuffersrc audio source buffer context
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* @param data pointers to the samples planes
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* @param linesize linesizes of each audio buffer plane
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* @param nb_samples number of samples per channel
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* @param sample_fmt sample format of the audio data
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* @param ch_layout channel layout of the audio data
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* @param planar flag to indicate if audio data is planar or packed
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* @param pts presentation timestamp of the audio buffer
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* @param flags unused
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*/
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int av_asrc_buffer_add_samples(AVFilterContext *abuffersrc,
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uint8_t *data[8], int linesize[8],
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int nb_samples, int sample_rate,
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int sample_fmt, int64_t ch_layout, int planar,
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int64_t pts, int av_unused flags);
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/**
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* Queue an audio buffer to the audio buffer source.
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*
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* This is similar to av_asrc_buffer_add_samples(), but the samples
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* are stored in a buffer with known size.
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*
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* @param abuffersrc audio source buffer context
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* @param buf pointer to the samples data, packed is assumed
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* @param size the size in bytes of the buffer, it must contain an
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* integer number of samples
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* @param sample_fmt sample format of the audio data
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* @param ch_layout channel layout of the audio data
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* @param pts presentation timestamp of the audio buffer
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* @param flags unused
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*/
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int av_asrc_buffer_add_buffer(AVFilterContext *abuffersrc,
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uint8_t *buf, int buf_size,
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int sample_rate,
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int sample_fmt, int64_t ch_layout, int planar,
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int64_t pts, int av_unused flags);
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/**
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* Queue an audio buffer to the audio buffer source.
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*
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* @param abuffersrc audio source buffer context
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* @param samplesref buffer ref to queue
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* @param flags unused
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*/
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int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *abuffersrc,
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AVFilterBufferRef *samplesref,
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int av_unused flags);
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#endif /* AVFILTER_ASRC_ABUFFER_H */
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