ffmpeg/libavfilter/asrc_anullsrc.c
Michael Niedermayer c7b9eab2be Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtmp: Add a new option 'rtmp_buffer', for setting the client buffer time
  rtmp: Set the client buffer time to 3s instead of 0.26s
  rtmp: Handle server bandwidth packets
  rtmp: Display a verbose message when an unknown packet type is received
  lavfi/audio: use av_samples_copy() instead of custom code.
  configure: add all filters hardcoded into avconv to avconv_deps
  avfiltergraph: remove a redundant call to avfilter_get_by_name().
  lavfi: allow building without swscale.
  build: Do not delete tests/vsynth2 directory, which is no longer created.
  lavfi: replace AVFilterContext.input/output_count with nb_inputs/outputs
  lavfi: make AVFilterPad opaque after two major bumps.
  lavfi: add avfilter_pad_get_type() and avfilter_pad_get_name().
  lavfi: make avfilter_get_video_buffer() private on next bump.
  jack: update to new latency range API as the old one has been deprecated
  rtmp: Tokenize the AMF connection parameters manually instead of using strtok_r
  ppc: Rename H.264 optimization template file for consistency.
  lavfi: add channelsplit audio filter.
  golomb: check remaining bits during unary decoding in get_ur_golomb_jpegls()
  sws: fix planar RGB input conversions for 9/10/16 bpp.

Conflicts:
	Changelog
	configure
	doc/APIchanges
	ffmpeg.c
	libavcodec/golomb.h
	libavcodec/v210dec.h
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/asrc_anullsrc.c
	libavfilter/audio.c
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/avfiltergraph.c
	libavfilter/buffersrc.c
	libavfilter/formats.c
	libavfilter/version.h
	libavfilter/vf_frei0r.c
	libavfilter/vf_pad.c
	libavfilter/vf_scale.c
	libavfilter/video.h
	libavfilter/vsrc_color.c
	libavformat/rtmpproto.c
	libswscale/input.c
	tests/Makefile

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-13 22:43:57 +02:00

139 lines
4.7 KiB
C

/*
* Copyright 2010 S.N. Hemanth Meenakshisundaram <smeenaks ucsd edu>
* Copyright 2010 Stefano Sabatini <stefano.sabatini-lala poste it>
*
* 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
* null audio source
*/
#include "internal.h"
#include "libavutil/audioconvert.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct {
const AVClass *class;
char *channel_layout_str;
uint64_t channel_layout;
char *sample_rate_str;
int sample_rate;
int nb_samples; ///< number of samples per requested frame
int64_t pts;
} ANullContext;
#define OFFSET(x) offsetof(ANullContext, x)
static const AVOption anullsrc_options[]= {
{ "channel_layout", "set channel_layout", OFFSET(channel_layout_str), AV_OPT_TYPE_STRING, {.str = "stereo"}, 0, 0 },
{ "cl", "set channel_layout", OFFSET(channel_layout_str), AV_OPT_TYPE_STRING, {.str = "stereo"}, 0, 0 },
{ "sample_rate", "set sample rate", OFFSET(sample_rate_str) , AV_OPT_TYPE_STRING, {.str = "44100"}, 0, 0 },
{ "r", "set sample rate", OFFSET(sample_rate_str) , AV_OPT_TYPE_STRING, {.str = "44100"}, 0, 0 },
{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.dbl = 1024}, 0, INT_MAX },
{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.dbl = 1024}, 0, INT_MAX },
{ NULL },
};
static const AVClass anullsrc_class = {
"ANullSrcContext",
av_default_item_name,
anullsrc_options
};
static int init(AVFilterContext *ctx, const char *args, void *opaque)
{
ANullContext *null = ctx->priv;
int ret;
null->class = &anullsrc_class;
av_opt_set_defaults(null);
if ((ret = (av_set_options_string(null, args, "=", ":"))) < 0) {
av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args);
return ret;
}
if ((ret = ff_parse_sample_rate(&null->sample_rate,
null->sample_rate_str, ctx)) < 0)
return ret;
if ((ret = ff_parse_channel_layout(&null->channel_layout,
null->channel_layout_str, ctx)) < 0)
return ret;
return 0;
}
static int config_props(AVFilterLink *outlink)
{
ANullContext *null = outlink->src->priv;
char buf[128];
int chans_nb;
outlink->sample_rate = null->sample_rate;
outlink->channel_layout = null->channel_layout;
chans_nb = av_get_channel_layout_nb_channels(null->channel_layout);
av_get_channel_layout_string(buf, sizeof(buf), chans_nb, null->channel_layout);
av_log(outlink->src, AV_LOG_INFO,
"sample_rate:%d channel_layout:'%s' nb_samples:%d\n",
null->sample_rate, buf, null->nb_samples);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
ANullContext *null = outlink->src->priv;
AVFilterBufferRef *samplesref;
samplesref =
ff_get_audio_buffer(outlink, AV_PERM_WRITE, null->nb_samples);
samplesref->pts = null->pts;
samplesref->pos = -1;
samplesref->audio->channel_layout = null->channel_layout;
samplesref->audio->sample_rate = outlink->sample_rate;
ff_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
avfilter_unref_buffer(samplesref);
null->pts += null->nb_samples;
return 0;
}
AVFilter avfilter_asrc_anullsrc = {
.name = "anullsrc",
.description = NULL_IF_CONFIG_SMALL("Null audio source, return empty audio frames."),
.init = init,
.priv_size = sizeof(ANullContext),
.inputs = (const AVFilterPad[]) {{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
.request_frame = request_frame, },
{ .name = NULL}},
};