Commit 587c8ab9 authored by Mina Nagy Zaki's avatar Mina Nagy Zaki Committed by Stefano Sabatini

lavfi: add asrc_abuffer - audio buffer source

Originally based on code by Stefano Sabatini and S. N. Hemanth.
Signed-off-by: 's avatarStefano Sabatini <stefano.sabatini-lala@poste.it>
parent f138c7f9
......@@ -1501,6 +1501,7 @@ tcp_protocol_deps="network"
udp_protocol_deps="network"
# filters
abuffer="strtok_r"
aformat_filter_deps="strtok_r"
blackframe_filter_deps="gpl"
boxblur_filter_deps="gpl"
......
......@@ -194,6 +194,51 @@ Adler-32 checksum for each input frame plane, expressed in the form
Below is a description of the currently available audio sources.
@section abuffer
Buffer audio frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular
through the interface defined in @file{libavfilter/asrc_abuffer.h}.
It accepts the following mandatory parameters:
@var{sample_rate}:@var{sample_fmt}:@var{channel_layout}:@var{packing}
@table @option
@item sample_rate
The sample rate of the incoming audio buffers.
@item sample_fmt
The sample format of the incoming audio buffers.
Either a sample format name or its corresponging integer representation from
the enum AVSampleFormat in @file{libavutil/samplefmt.h}
@item channel_layout
The channel layout of the incoming audio buffers.
Either a channel layout name from channel_layout_map in
@file{libavutil/audioconvert.c} or its corresponding integer representation
from the AV_CH_LAYOUT_* macros in @file{libavutil/audioconvert.h}
@item packing
Either "packed" or "planar", or their integer representation: 0 or 1
respectively.
@end table
For example:
@example
abuffer=44100:s16:stereo:planar
@end example
will instruct the source to accept planar 16bit signed stereo at 44100Hz.
Since the sample format with name "s16" corresponds to the number
1 and the "stereo" channel layout corresponds to the value 3, this is
equivalent to:
@example
abuffer=44100:1:3:1
@end example
@section anullsrc
Null audio source, never return audio frames. It is mainly useful as a
......
......@@ -24,6 +24,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
OBJS-$(CONFIG_ABUFFERSINK_FILTER) += asink_abuffer.o
......
......@@ -39,6 +39,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
REGISTER_FILTER (ABUFFER, abuffer, asrc);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
REGISTER_FILTER (ABUFFERSINK, abuffersink, asink);
......
This diff is collapsed.
/*
* 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
*/
#ifndef AVFILTER_ASRC_ABUFFER_H
#define AVFILTER_ASRC_ABUFFER_H
#include "avfilter.h"
/**
* @file
* memory buffer source for audio
*/
/**
* Queue an audio buffer to the audio buffer source.
*
* @param abuffersrc audio source buffer context
* @param data pointers to the samples planes
* @param linesize linesizes of each audio buffer plane
* @param nb_samples number of samples per channel
* @param sample_fmt sample format of the audio data
* @param ch_layout channel layout of the audio data
* @param planar flag to indicate if audio data is planar or packed
* @param pts presentation timestamp of the audio buffer
* @param flags unused
*/
int av_asrc_buffer_add_samples(AVFilterContext *abuffersrc,
uint8_t *data[8], int linesize[8],
int nb_samples, int sample_rate,
int sample_fmt, int64_t ch_layout, int planar,
int64_t pts, int av_unused flags);
/**
* Queue an audio buffer to the audio buffer source.
*
* This is similar to av_asrc_buffer_add_samples(), but the samples
* are stored in a buffer with known size.
*
* @param abuffersrc audio source buffer context
* @param buf pointer to the samples data, packed is assumed
* @param size the size in bytes of the buffer, it must contain an
* integer number of samples
* @param sample_fmt sample format of the audio data
* @param ch_layout channel layout of the audio data
* @param pts presentation timestamp of the audio buffer
* @param flags unused
*/
int av_asrc_buffer_add_buffer(AVFilterContext *abuffersrc,
uint8_t *buf, int buf_size,
int sample_rate,
int sample_fmt, int64_t ch_layout, int planar,
int64_t pts, int av_unused flags);
/**
* Queue an audio buffer to the audio buffer source.
*
* @param abuffersrc audio source buffer context
* @param samplesref buffer ref to queue
* @param flags unused
*/
int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *abuffersrc,
AVFilterBufferRef *samplesref,
int av_unused flags);
#endif /* AVFILTER_ASRC_ABUFFER_H */
......@@ -29,7 +29,7 @@
#include "libavutil/rational.h"
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 33
#define LIBAVFILTER_VERSION_MINOR 34
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
......
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