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lavfi: merge asrc_abuffer into vsrc_buffer.

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This commit is contained in:
Nicolas George 2012-04-27 19:19:45 +02:00
parent 28c5775183
commit 7ad0d9864e
3 changed files with 339 additions and 382 deletions
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1
libavfilter/Makefile
View File

@ -42,7 +42,6 @@ OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o
OBJS-$(CONFIG_AMOVIE_FILTER) += src_movie.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o

369
libavfilter/asrc_abuffer.c
View File

@ -1,369 +0,0 @@
/*
* 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/avstring.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:%d -> %s:%s:%d, normalizing\n",
av_get_sample_fmt_name(link->format),
old_layout_str, (int)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] = {0};
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, 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 *args0, void *opaque)
{
ABufferSourceContext *abuffer = ctx->priv;
char *arg = NULL, *ptr, chlayout_str[16];
char *args = av_strdup(args0);
int ret;
arg = av_strtok(args, ":", &ptr);
#define ADD_FORMAT(fmt_name) \
if (!arg) \
goto arg_fail; \
if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) { \
av_freep(&args); \
return ret; \
} \
if (*args) \
arg = av_strtok(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);
av_freep(&args);
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");
av_freep(&args);
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))
return AVERROR(EAGAIN);
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 = (const AVFilterPad[]) {{ .name = NULL }},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
.poll_frame = poll_frame,
.config_props = config_output, },
{ .name = NULL}},
};

351
libavfilter/vsrc_buffer.c
View File

@ -1,5 +1,7 @@
/*
* Copyright (c) 2008 Vitor Sessak
* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram
* Copyright (c) 2011 Mina Nagy Zaki
*
* This file is part of FFmpeg.
*
@ -28,21 +30,39 @@
#include "avcodec.h"
#include "buffersrc.h"
#include "vsrc_buffer.h"
#include "asrc_abuffer.h"
#include "libavutil/audioconvert.h"
#include "libavutil/avstring.h"
#include "libavutil/fifo.h"
#include "libavutil/imgutils.h"
typedef struct {
AVFilterContext *scale;
AVFifoBuffer *fifo;
int h, w;
enum PixelFormat pix_fmt;
AVRational time_base; ///< time_base to set in the output link
AVRational sample_aspect_ratio;
char sws_param[256];
int eof;
unsigned nb_failed_requests;
/* Video only */
AVFilterContext *scale;
int h, w;
enum PixelFormat pix_fmt;
AVRational sample_aspect_ratio;
char sws_param[256];
/* Audio only */
// Audio format of incoming buffers
int sample_rate;
unsigned int sample_format;
int64_t channel_layout;
int packing_format;
// Normalization filters
AVFilterContext *aconvert;
AVFilterContext *aresample;
} BufferSourceContext;
#define FIFO_SIZE 8
#define CHECK_PARAM_CHANGE(s, c, width, height, format)\
if (c->w != width || c->h != height || c->pix_fmt != format) {\
av_log(s, AV_LOG_ERROR, "Changing frame properties on the fly is not supported.\n");\
@ -186,7 +206,7 @@ unsigned av_vsrc_buffer_get_nb_failed_requests(AVFilterContext *buffer_src)
return ((BufferSourceContext *)buffer_src->priv)->nb_failed_requests;
}
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
static av_cold int init_video(AVFilterContext *ctx, const char *args, void *opaque)
{
BufferSourceContext *c = ctx->priv;
char pix_fmt_str[128];
@ -214,6 +234,52 @@ static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
return 0;
}
static av_cold int init_audio(AVFilterContext *ctx, const char *args0, void *opaque)
{
BufferSourceContext *abuffer = ctx->priv;
char *arg = NULL, *ptr, chlayout_str[16];
char *args = av_strdup(args0);
int ret;
arg = av_strtok(args, ":", &ptr);
#define ADD_FORMAT(fmt_name) \
if (!arg) \
goto arg_fail; \
if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) { \
av_freep(&args); \
return ret; \
} \
if (*args) \
arg = av_strtok(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);
av_freep(&args);
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");
av_freep(&args);
return AVERROR(EINVAL);
}
static av_cold void uninit(AVFilterContext *ctx)
{
BufferSourceContext *s = ctx->priv;
@ -228,7 +294,7 @@ static av_cold void uninit(AVFilterContext *ctx)
s->scale = NULL;
}
static int query_formats(AVFilterContext *ctx)
static int query_formats_video(AVFilterContext *ctx)
{
BufferSourceContext *c = ctx->priv;
enum PixelFormat pix_fmts[] = { c->pix_fmt, PIX_FMT_NONE };
@ -237,7 +303,27 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
static int config_props(AVFilterLink *link)
static int query_formats_audio(AVFilterContext *ctx)
{
BufferSourceContext *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_video(AVFilterLink *link)
{
BufferSourceContext *c = link->src->priv;
@ -249,6 +335,13 @@ static int config_props(AVFilterLink *link)
return 0;
}
static int config_output_audio(AVFilterLink *outlink)
{
BufferSourceContext *abuffer = outlink->src->priv;
outlink->sample_rate = abuffer->sample_rate;
return 0;
}
static int request_frame(AVFilterLink *link)
{
BufferSourceContext *c = link->src->priv;
@ -262,11 +355,20 @@ static int request_frame(AVFilterLink *link)
}
av_fifo_generic_read(c->fifo, &buf, sizeof(buf), NULL);
switch (link->type) {
case AVMEDIA_TYPE_VIDEO:
/* TODO reindent */
avfilter_start_frame(link, avfilter_ref_buffer(buf, ~0));
avfilter_draw_slice(link, 0, link->h, 1);
avfilter_end_frame(link);
avfilter_unref_buffer(buf);
break;
case AVMEDIA_TYPE_AUDIO:
avfilter_filter_samples(link, avfilter_ref_buffer(buf, ~0));
break;
default:
return AVERROR(ENOSYS);
}
return 0;
}
@ -279,13 +381,216 @@ static int poll_frame(AVFilterLink *link)
return size/sizeof(AVFilterBufferRef*);
}
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(BufferSourceContext *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(BufferSourceContext *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:%d -> %s:%s:%d, normalizing\n",
av_get_sample_fmt_name(link->format),
old_layout_str, (int)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)
{
BufferSourceContext *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] = {0};
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, 16);
return av_asrc_buffer_add_samples(ctx,
data, linesize, nb_samples,
sample_rate,
sample_fmt, channel_layout, planar,
pts, flags);
}
AVFilter avfilter_vsrc_buffer = {
.name = "buffer",
.description = NULL_IF_CONFIG_SMALL("Buffer video frames, and make them accessible to the filterchain."),
.priv_size = sizeof(BufferSourceContext),
.query_formats = query_formats,
.query_formats = query_formats_video,
.init = init,
.init = init_video,
.uninit = uninit,
.inputs = (const AVFilterPad[]) {{ .name = NULL }},
@ -293,6 +598,28 @@ AVFilter avfilter_vsrc_buffer = {
.type = AVMEDIA_TYPE_VIDEO,
.request_frame = request_frame,
.poll_frame = poll_frame,
.config_props = config_props, },
.config_props = config_output_video, },
{ .name = NULL}},
};
#ifdef CONFIG_ABUFFER_FILTER
AVFilter avfilter_asrc_abuffer = {
.name = "abuffer",
.description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."),
.priv_size = sizeof(BufferSourceContext),
.query_formats = query_formats_audio,
.init = init_audio,
.uninit = uninit,
.inputs = (const AVFilterPad[]) {{ .name = NULL }},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
.poll_frame = poll_frame,
.config_props = config_output_audio, },
{ .name = NULL}},
};
#endif