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lavfi: add asrc_abuffer - audio buffer source

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Originally based on code by Stefano Sabatini and S. N. Hemanth.

Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
This commit is contained in:
Mina Nagy Zaki
2011-08-01 11:33:26 +03:00
committed by Stefano Sabatini
parent f138c7f993
commit 587c8ab912
7 changed files with 495 additions and 1 deletions
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1
configure vendored
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@@ -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"

45
doc/filters.texi
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@@ -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

1
libavfilter/Makefile
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@@ -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

1
libavfilter/allfilters.c
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@@ -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);

366
libavfilter/asrc_abuffer.c Normal file
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@@ -0,0 +1,366 @@
/*
* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram
* Copyright (c) 2011 Mina Nagy Zaki
*
* 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
*/
/**
* @file
* memory buffer source for audio
*/
#include "libavutil/audioconvert.h"
#include "libavutil/fifo.h"
#include "asrc_abuffer.h"
#include "internal.h"
typedef struct {
// Audio format of incoming buffers
int sample_rate;
unsigned int sample_format;
int64_t channel_layout;
int packing_format;
// FIFO buffer of audio buffer ref pointers
AVFifoBuffer *fifo;
// Normalization filters
AVFilterContext *aconvert;
AVFilterContext *aresample;
} ABufferSourceContext;
#define FIFO_SIZE 8
static void buf_free(AVFilterBuffer *ptr)
{
av_free(ptr);
return;
}
static void set_link_source(AVFilterContext *src, AVFilterLink *link)
{
link->src = src;
link->srcpad = &(src->output_pads[0]);
src->outputs[0] = link;
}
static int reconfigure_filter(ABufferSourceContext *abuffer, AVFilterContext *filt_ctx)
{
int ret;
AVFilterLink * const inlink = filt_ctx->inputs[0];
AVFilterLink * const outlink = filt_ctx->outputs[0];
inlink->format = abuffer->sample_format;
inlink->channel_layout = abuffer->channel_layout;
inlink->planar = abuffer->packing_format;
inlink->sample_rate = abuffer->sample_rate;
filt_ctx->filter->uninit(filt_ctx);
memset(filt_ctx->priv, 0, filt_ctx->filter->priv_size);
if ((ret = filt_ctx->filter->init(filt_ctx, NULL , NULL)) < 0)
return ret;
if ((ret = inlink->srcpad->config_props(inlink)) < 0)
return ret;
return outlink->srcpad->config_props(outlink);
}
static int insert_filter(ABufferSourceContext *abuffer,
AVFilterLink *link, AVFilterContext **filt_ctx,
const char *filt_name)
{
int ret;
if ((ret = avfilter_open(filt_ctx, avfilter_get_by_name(filt_name), NULL)) < 0)
return ret;
link->src->outputs[0] = NULL;
if ((ret = avfilter_link(link->src, 0, *filt_ctx, 0)) < 0) {
link->src->outputs[0] = link;
return ret;
}
set_link_source(*filt_ctx, link);
if ((ret = reconfigure_filter(abuffer, *filt_ctx)) < 0) {
avfilter_free(*filt_ctx);
return ret;
}
return 0;
}
static void remove_filter(AVFilterContext **filt_ctx)
{
AVFilterLink *outlink = (*filt_ctx)->outputs[0];
AVFilterContext *src = (*filt_ctx)->inputs[0]->src;
(*filt_ctx)->outputs[0] = NULL;
avfilter_free(*filt_ctx);
*filt_ctx = NULL;
set_link_source(src, outlink);
}
static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBufferRef *ref)
{
char old_layout_str[16], new_layout_str[16];
av_get_channel_layout_string(old_layout_str, sizeof(old_layout_str),
-1, link->channel_layout);
av_get_channel_layout_string(new_layout_str, sizeof(new_layout_str),
-1, ref->audio->channel_layout);
av_log(ctx, AV_LOG_INFO,
"Audio input format changed: "
"%s:%s:%"PRId64" -> %s:%s:%u, normalizing\n",
av_get_sample_fmt_name(link->format),
old_layout_str, link->sample_rate,
av_get_sample_fmt_name(ref->format),
new_layout_str, ref->audio->sample_rate);
}
int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *ctx,
AVFilterBufferRef *samplesref,
int av_unused flags)
{
ABufferSourceContext *abuffer = ctx->priv;
AVFilterLink *link;
int ret, logged = 0;
if (av_fifo_space(abuffer->fifo) < sizeof(samplesref)) {
av_log(ctx, AV_LOG_ERROR,
"Buffering limit reached. Please consume some available frames "
"before adding new ones.\n");
return AVERROR(EINVAL);
}
// Normalize input
link = ctx->outputs[0];
if (samplesref->audio->sample_rate != link->sample_rate) {
log_input_change(ctx, link, samplesref);
logged = 1;
abuffer->sample_rate = samplesref->audio->sample_rate;
if (!abuffer->aresample) {
ret = insert_filter(abuffer, link, &abuffer->aresample, "aresample");
if (ret < 0) return ret;
} else {
link = abuffer->aresample->outputs[0];
if (samplesref->audio->sample_rate == link->sample_rate)
remove_filter(&abuffer->aresample);
else
if ((ret = reconfigure_filter(abuffer, abuffer->aresample)) < 0)
return ret;
}
}
link = ctx->outputs[0];
if (samplesref->format != link->format ||
samplesref->audio->channel_layout != link->channel_layout ||
samplesref->audio->planar != link->planar) {
if (!logged) log_input_change(ctx, link, samplesref);
abuffer->sample_format = samplesref->format;
abuffer->channel_layout = samplesref->audio->channel_layout;
abuffer->packing_format = samplesref->audio->planar;
if (!abuffer->aconvert) {
ret = insert_filter(abuffer, link, &abuffer->aconvert, "aconvert");
if (ret < 0) return ret;
} else {
link = abuffer->aconvert->outputs[0];
if (samplesref->format == link->format &&
samplesref->audio->channel_layout == link->channel_layout &&
samplesref->audio->planar == link->planar
)
remove_filter(&abuffer->aconvert);
else
if ((ret = reconfigure_filter(abuffer, abuffer->aconvert)) < 0)
return ret;
}
}
if (sizeof(samplesref) != av_fifo_generic_write(abuffer->fifo, &samplesref,
sizeof(samplesref), NULL)) {
av_log(ctx, AV_LOG_ERROR, "Error while writing to FIFO\n");
return AVERROR(EINVAL);
}
return 0;
}
int av_asrc_buffer_add_samples(AVFilterContext *ctx,
uint8_t *data[8], int linesize[8],
int nb_samples, int sample_rate,
int sample_fmt, int64_t channel_layout, int planar,
int64_t pts, int av_unused flags)
{
AVFilterBufferRef *samplesref;
samplesref = avfilter_get_audio_buffer_ref_from_arrays(
data, linesize, AV_PERM_WRITE,
nb_samples,
sample_fmt, channel_layout, planar);
if (!samplesref)
return AVERROR(ENOMEM);
samplesref->buf->free = buf_free;
samplesref->pts = pts;
samplesref->audio->sample_rate = sample_rate;
return av_asrc_buffer_add_audio_buffer_ref(ctx, samplesref, 0);
}
int av_asrc_buffer_add_buffer(AVFilterContext *ctx,
uint8_t *buf, int buf_size, int sample_rate,
int sample_fmt, int64_t channel_layout, int planar,
int64_t pts, int av_unused flags)
{
uint8_t *data[8];
int linesize[8];
int nb_channels = av_get_channel_layout_nb_channels(channel_layout),
nb_samples = buf_size / nb_channels / av_get_bytes_per_sample(sample_fmt);
av_samples_fill_arrays(data, linesize,
buf, nb_channels, nb_samples,
sample_fmt, planar, 16);
return av_asrc_buffer_add_samples(ctx,
data, linesize, nb_samples,
sample_rate,
sample_fmt, channel_layout, planar,
pts, flags);
}
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
{
ABufferSourceContext *abuffer = ctx->priv;
char *arg = NULL, *ptr, chlayout_str[16];
int ret;
arg = strtok_r(args, ":", &ptr);
#define ADD_FORMAT(fmt_name) \
if (!arg) \
goto arg_fail; \
if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) \
return ret; \
if (*args) \
arg = strtok_r(NULL, ":", &ptr)
ADD_FORMAT(sample_rate);
ADD_FORMAT(sample_format);
ADD_FORMAT(channel_layout);
ADD_FORMAT(packing_format);
abuffer->fifo = av_fifo_alloc(FIFO_SIZE*sizeof(AVFilterBufferRef*));
if (!abuffer->fifo) {
av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo, filter init failed.\n");
return AVERROR(ENOMEM);
}
av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str),
-1, abuffer->channel_layout);
av_log(ctx, AV_LOG_INFO, "format:%s layout:%s rate:%d\n",
av_get_sample_fmt_name(abuffer->sample_format), chlayout_str,
abuffer->sample_rate);
return 0;
arg_fail:
av_log(ctx, AV_LOG_ERROR, "Invalid arguments, must be of the form "
"sample_rate:sample_fmt:channel_layout:packing\n");
return AVERROR(EINVAL);
}
static av_cold void uninit(AVFilterContext *ctx)
{
ABufferSourceContext *abuffer = ctx->priv;
av_fifo_free(abuffer->fifo);
}
static int query_formats(AVFilterContext *ctx)
{
ABufferSourceContext *abuffer = ctx->priv;
AVFilterFormats *formats;
formats = NULL;
avfilter_add_format(&formats, abuffer->sample_format);
avfilter_set_common_sample_formats(ctx, formats);
formats = NULL;
avfilter_add_format(&formats, abuffer->channel_layout);
avfilter_set_common_channel_layouts(ctx, formats);
formats = NULL;
avfilter_add_format(&formats, abuffer->packing_format);
avfilter_set_common_packing_formats(ctx, formats);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
ABufferSourceContext *abuffer = outlink->src->priv;
outlink->sample_rate = abuffer->sample_rate;
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
ABufferSourceContext *abuffer = outlink->src->priv;
AVFilterBufferRef *samplesref;
if (!av_fifo_size(abuffer->fifo)) {
av_log(outlink->src, AV_LOG_ERROR,
"request_frame() called with no available frames!\n");
return AVERROR(EINVAL);
}
av_fifo_generic_read(abuffer->fifo, &samplesref, sizeof(samplesref), NULL);
avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
avfilter_unref_buffer(samplesref);
return 0;
}
static int poll_frame(AVFilterLink *outlink)
{
ABufferSourceContext *abuffer = outlink->src->priv;
return av_fifo_size(abuffer->fifo)/sizeof(AVFilterBufferRef*);
}
AVFilter avfilter_asrc_abuffer = {
.name = "abuffer",
.description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."),
.priv_size = sizeof(ABufferSourceContext),
.query_formats = query_formats,
.init = init,
.uninit = uninit,
.inputs = (AVFilterPad[]) {{ .name = NULL }},
.outputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
.poll_frame = poll_frame,
.config_props = config_output, },
{ .name = NULL}},
};

80
libavfilter/asrc_abuffer.h Normal file
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@@ -0,0 +1,80 @@
/*
* 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 */

2
libavfilter/avfilter.h
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@@ -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, \