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asrc_anullsrc: implement a request_frame callback for returning frames
This is mainly useful for filters (like the sox synth), which overwrite the content of the passed data.
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@ -309,8 +309,10 @@ value is "-1".
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@section anullsrc
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Null audio source, never return audio frames. It is mainly useful as a
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template and to be employed in analysis / debugging tools.
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Null audio source, return unprocessed audio frames. It is mainly useful
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as a template and to be employed in analysis / debugging tools, or as
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the source for filters which ignore the input data (for example the sox
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synth filter).
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It accepts an optional sequence of @var{key}=@var{value} pairs,
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separated by ":".
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@ -331,6 +333,10 @@ is "stereo".
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Check the channel_layout_map definition in
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@file{libavcodec/audioconvert.c} for the mapping between strings and
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channel layout values.
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@item nb_samples, n
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Set the number of samples per requested frames.
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@end table
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Follow some examples:
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@ -33,6 +33,8 @@ typedef struct {
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int64_t channel_layout;
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char *sample_rate_str;
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int sample_rate;
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int nb_samples; ///< number of samples per requested frame
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int64_t pts;
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} ANullContext;
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#define OFFSET(x) offsetof(ANullContext, x)
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@ -42,6 +44,8 @@ static const AVOption anullsrc_options[]= {
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{ "cl", "set channel_layout", OFFSET(channel_layout_str), FF_OPT_TYPE_STRING, {.str = "stereo"}, 0, 0 },
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{ "sample_rate", "set sample rate", OFFSET(sample_rate_str) , FF_OPT_TYPE_STRING, {.str = "44100"}, 0, 0 },
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{ "r", "set sample rate", OFFSET(sample_rate_str) , FF_OPT_TYPE_STRING, {.str = "44100"}, 0, 0 },
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{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), FF_OPT_TYPE_INT, {.dbl = 1024}, 0, INT_MAX },
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{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), FF_OPT_TYPE_INT, {.dbl = 1024}, 0, INT_MAX },
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{ NULL },
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};
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@ -92,15 +96,29 @@ static int config_props(AVFilterLink *outlink)
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chans_nb = av_get_channel_layout_nb_channels(priv->channel_layout);
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av_get_channel_layout_string(buf, sizeof(buf), chans_nb, priv->channel_layout);
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av_log(outlink->src, AV_LOG_INFO,
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"sample_rate:%d channel_layout:%"PRId64 " channel_layout_description:'%s'\n",
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priv->sample_rate, priv->channel_layout, buf);
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"sample_rate:%d channel_layout:%"PRId64 " channel_layout_description:'%s' nb_samples:%d\n",
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priv->sample_rate, priv->channel_layout, buf, priv->nb_samples);
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return 0;
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}
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static int request_frame(AVFilterLink *link)
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static int request_frame(AVFilterLink *outlink)
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{
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return -1;
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ANullContext *null = outlink->src->priv;
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AVFilterBufferRef *samplesref;
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samplesref =
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avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, null->nb_samples);
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samplesref->pts = null->pts;
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samplesref->pos = -1;
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samplesref->audio->channel_layout = null->channel_layout;
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samplesref->audio->sample_rate = outlink->sample_rate;
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avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
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avfilter_unref_buffer(samplesref);
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null->pts += null->nb_samples;
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return 0;
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}
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AVFilter avfilter_asrc_anullsrc = {
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@ -30,7 +30,7 @@
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#define LIBAVFILTER_VERSION_MAJOR 2
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#define LIBAVFILTER_VERSION_MINOR 43
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#define LIBAVFILTER_VERSION_MICRO 1
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#define LIBAVFILTER_VERSION_MICRO 2
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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LIBAVFILTER_VERSION_MINOR, \
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