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FFmpeg/libavfilter/asrc_anullsrc.c
Michael Niedermayer 015903294c Merge remote-tracking branch 'qatar/master'
* qatar/master: (25 commits)
  rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC
  ape: Use unsigned integer maths
  arm: dsputil: fix overreads in put/avg_pixels functions
  h264: K&R formatting cosmetics for header files (part II/II)
  h264: K&R formatting cosmetics for header files (part I/II)
  rtmp: Implement check bandwidth notification.
  rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player.
  rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin.
  rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream.
  cmdutils: Add fallback case to switch in check_stream_specifier().
  sctp: be consistent with socket option level
  configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags.
  vcr1enc: drop pointless empty encode_init() wrapper function
  vcr1: drop pointless write-only AVCodecContext member from VCR1Context
  vcr1: group encoder code together to save #ifdefs
  vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments
  mov: make one comment slightly more specific
  lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX
  lavfi: move audio-related functions to a separate file.
  lavfi: remove some audio-related function from public API.
  ...

Conflicts:
	cmdutils.c
	libavcodec/h264.h
	libavcodec/h264_mvpred.h
	libavcodec/vcr1.c
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/defaults.c
	libavfilter/internal.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-10 23:30:42 +02:00

139 lines
4.6 KiB
C

/*
* 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 "libavutil/audioconvert.h"
#include "libavutil/opt.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 char *anullsrc_get_name(void *ctx)
{
return "anullsrc";
}
static const AVClass anullsrc_class = {
"ANullSrcContext",
anullsrc_get_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}},
};