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diff --git a/doc/filters.texi b/doc/filters.texi index 2af7b37d07..ba8c9f46fb 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -211,6 +211,32 @@ amovie=input.mkv:si=5 [a5]; [x3][a5] amerge" -c:a pcm_s16le output.mkv @end example +@section aformat + +Convert the input audio to one of the specified formats. The framework will +negotiate the most appropriate format to minimize conversions. + +The filter accepts the following named parameters: +@table @option + +@item sample_fmts +A comma-separated list of requested sample formats. + +@item sample_rates +A comma-separated list of requested sample rates. + +@item channel_layouts +A comma-separated list of requested channel layouts. + +@end table + +If a parameter is omitted, all values are allowed. + +For example to force the output to either unsigned 8-bit or signed 16-bit stereo: +@example +aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo +@end example + @section anull Pass the audio source unchanged to the output. @@ -502,6 +528,25 @@ volume=-12dB @end example @end itemize +@section asyncts +Synchronize audio data with timestamps by squeezing/stretching it and/or +dropping samples/adding silence when needed. + +The filter accepts the following named parameters: +@table @option + +@item compensate +Enable stretching/squeezing the data to make it match the timestamps. + +@item min_delta +Minimum difference between timestamps and audio data (in seconds) to trigger +adding/dropping samples. + +@item max_comp +Maximum compensation in samples per second. + +@end table + @section resample Convert the audio sample format, sample rate and channel layout. This filter is not meant to be used directly. @@ -721,6 +766,33 @@ anullsrc=r=48000:cl=4 anullsrc=r=48000:cl=mono @end example +@section abuffer +Buffer audio frames, and make them available to the filter chain. + +This source is not intended to be part of user-supplied graph descriptions but +for insertion by calling programs through the interface defined in +@file{libavfilter/buffersrc.h}. + +It accepts the following named parameters: +@table @option + +@item time_base +Timebase which will be used for timestamps of submitted frames. It must be +either a floating-point number or in @var{numerator}/@var{denominator} form. + +@item sample_rate +Audio sample rate. + +@item sample_fmt +Name of the sample format, as returned by @code{av_get_sample_fmt_name()}. + +@item channel_layout +Channel layout of the audio data, in the form that can be accepted by +@code{av_get_channel_layout()}. +@end table + +All the parameters need to be explicitly defined. + @c man end AUDIO SOURCES @chapter Audio Sinks @@ -745,6 +817,13 @@ Null audio sink, do absolutely nothing with the input audio. It is mainly useful as a template and to be employed in analysis / debugging tools. +@section abuffersink +This sink is intended for programmatic use. Frames that arrive on this sink can +be retrieved by the calling program using the interface defined in +@file{libavfilter/buffersink.h}. + +This filter accepts no parameters. + @c man end AUDIO SINKS @chapter Video Filters |