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authorPaul B Mahol <onemda@gmail.com>2015-11-25 11:36:45 +0100
committerPaul B Mahol <onemda@gmail.com>2015-11-28 17:56:40 +0100
commit1685a781cd50dbc1c9fd3107ba57981ba452b127 (patch)
tree2867cf79677bad6f1ca3dfac9928493bc57ef6d6 /doc
parent3f895dcb0dcbcbf10a621bf4bfae6d8879899015 (diff)
avfilter: add audio compressor filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
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diff --git a/doc/filters.texi b/doc/filters.texi
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@@ -318,6 +318,78 @@ build.
Below is a description of the currently available audio filters.
+@section acompressor
+
+A compressor is mainly used to reduce the dynamic range of a signal.
+Especially modern music is mostly compressed at a high ratio to
+improve the overall loudness. It's done to get the highest attention
+of a listener, "fatten" the sound and bring more "power" to the track.
+If a signal is compressed too much it may sound dull or "dead"
+afterwards or it may start to "pump" (which could be a powerful effect
+but can also destroy a track completely).
+The right compression is the key to reach a professional sound and is
+the high art of mixing and mastering. Because of its complex settings
+it may take a long time to get the right feeling for this kind of effect.
+
+Compression is done by detecting the volume above a chosen level
+@code{threshold} and dividing it by the factor set with @code{ratio}.
+So if you set the threshold to -12dB and your signal reaches -6dB a ratio
+of 2:1 will result in a signal at -9dB. Because an exact manipulation of
+the signal would cause distortion of the waveform the reduction can be
+levelled over the time. This is done by setting "Attack" and "Release".
+@code{attack} determines how long the signal has to rise above the threshold
+before any reduction will occur and @code{release} sets the time the signal
+has to fall below the threshold to reduce the reduction again. Shorter signals
+than the chosen attack time will be left untouched.
+The overall reduction of the signal can be made up afterwards with the
+@code{makeup} setting. So compressing the peaks of a signal about 6dB and
+raising the makeup to this level results in a signal twice as loud than the
+source. To gain a softer entry in the compression the @code{knee} flattens the
+hard edge at the threshold in the range of the chosen decibels.
+
+The filter accepts the following options:
+
+@table @option
+@item threshold
+If a signal of second stream rises above this level it will affect the gain
+reduction of the first stream.
+By default it is 0.125. Range is between 0.00097563 and 1.
+
+@item ratio
+Set a ratio by which the signal is reduced. 1:2 means that if the level
+rose 4dB above the threshold, it will be only 2dB above after the reduction.
+Default is 2. Range is between 1 and 20.
+
+@item attack
+Amount of milliseconds the signal has to rise above the threshold before gain
+reduction starts. Default is 20. Range is between 0.01 and 2000.
+
+@item release
+Amount of milliseconds the signal has to fall below the threshold before
+reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
+
+@item makeup
+Set the amount by how much signal will be amplified after processing.
+Default is 2. Range is from 1 and 64.
+
+@item knee
+Curve the sharp knee around the threshold to enter gain reduction more softly.
+Default is 2.82843. Range is between 1 and 8.
+
+@item link
+Choose if the @code{average} level between all channels of input stream
+or the louder(@code{maximum}) channel of input stream affects the
+reduction. Default is @code{average}.
+
+@item detection
+Should the exact signal be taken in case of @code{peak} or an RMS one in case
+of @code{rms}. Default is @code{rms} which is mostly smoother.
+
+@item mix
+How much to use compressed signal in output. Default is 1.
+Range is between 0 and 1.
+@end table
+
@section acrossfade
Apply cross fade from one input audio stream to another input audio stream.