RepoMirrors/ffmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-25 08:42:39 +00:00
Code Issues Projects Releases Wiki Activity

avfilter/asrc_anullsrc: add support to set output duration

Browse Source
This commit is contained in:
Paul B Mahol 2020-09-04 18:30:46 +02:00
parent 88db1745fc
commit 615d75f291
2 changed files with 14 additions and 0 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

7
doc/filters.texi
View File

@ -6076,6 +6076,13 @@ Specifies the sample rate, and defaults to 44100.
@item nb_samples, n
Set the number of samples per requested frames.
@item duration, d
Set the duration of the sourced audio. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
If not specified, or the expressed duration is negative, the audio is
supposed to be generated forever.
@end table
@subsection Examples

7
libavfilter/asrc_anullsrc.c
View File

@ -40,6 +40,7 @@ typedef struct ANullContext {
uint64_t channel_layout;
char *sample_rate_str;
int sample_rate;
int64_t duration;
int nb_samples; ///< number of samples per requested frame
int64_t pts;
} ANullContext;
@ -54,6 +55,8 @@ static const AVOption anullsrc_options[]= {
{ "r", "set sample rate", OFFSET(sample_rate_str) , AV_OPT_TYPE_STRING, {.str = "44100"}, 0, 0, FLAGS },
{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 0, INT_MAX, FLAGS },
{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 0, INT_MAX, FLAGS },
{ "duration", "set the audio duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = -1}, -1, INT64_MAX, FLAGS },
{ "d", "set the audio duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = -1}, -1, INT64_MAX, FLAGS },
{ NULL }
};
@ -108,6 +111,10 @@ static int request_frame(AVFilterLink *outlink)
ANullContext *null = outlink->src->priv;
AVFrame *samplesref;
if (null->duration >= 0 &&
av_rescale_q(null->pts, outlink->time_base, AV_TIME_BASE_Q) >= null->duration)
return AVERROR_EOF;
samplesref = ff_get_audio_buffer(outlink, null->nb_samples);
if (!samplesref)
return AVERROR(ENOMEM);