Merge commit 'bd31c61cf94d01dbe1051cf65874e7b2c0ac5454'

[ffmpeg] / doc / filters.texi

diff --git a/doc/filters.texi b/doc/filters.texi
index 9fc8c4f908e53fe10d170a033f3b5d5a3bfc49ec..d1ee6f4c3e9c1f4e5f8b0563b8d70292eed3e6fc 100644 (file)
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -449,7 +449,7 @@ This filter is bit crusher with enhanced functionality. A bit crusher
 is used to audibly reduce number of bits an audio signal is sampled
 with. This doesn't change the bit depth at all, it just produces the
 effect. Material reduced in bit depth sounds more harsh and "digital".
-This filter is able to even round to continous values instead of discrete
+This filter is able to even round to continuous values instead of discrete
 bit depths.
 Additionally it has a D/C offset which results in different crushing of
 the lower and the upper half of the signal.
@@ -475,7 +475,7 @@ Set level out.
 Set bit reduction.
 
 @item mix
-Set mixing ammount.
+Set mixing amount.
 
 @item mode
 Can be linear: @code{lin} or logarithmic: @code{log}.
@@ -1203,7 +1203,7 @@ Set video stream size. Only useful if curves option is activated.
 
 @item mgain
 Set max gain that will be displayed. Only useful if curves option is activated.
-Setting this to reasonable value allows to display gain which is derived from
+Setting this to a reasonable value makes it possible to display gain which is derived from
 neighbour bands which are too close to each other and thus produce higher gain
 when both are activated.
 
@@ -2508,6 +2508,8 @@ and functions:
 @table @option
 @item gain_interpolate(f)
 interpolate gain on frequency f based on gain_entry
+@item cubic_interpolate(f)
+same as gain_interpolate, but smoother
 @end table
 This option is also available as command. Default is @code{gain_interpolate(f)}.
 
@@ -2583,6 +2585,10 @@ Set file for dumping, suitable for gnuplot.
 @item dumpscale
 Set scale for dumpfile. Acceptable values are same with scale option.
 Default is linlog.
+
+@item fft2
+Enable 2-channel convolution using complex FFT. This improves speed significantly.
+Default is disabled.
 @end table
 
 @subsection Examples
@@ -2915,9 +2921,6 @@ EBU R128 loudness normalization. Includes both dynamic and linear normalization
 Support for both single pass (livestreams, files) and double pass (files) modes.
 This algorithm can target IL, LRA, and maximum true peak.
 
-To enable compilation of this filter you need to configure FFmpeg with
-@code{--enable-libebur128}.
-
 The filter accepts the following options:
 
 @table @option
@@ -8856,7 +8859,7 @@ value.
 @section hysteresis
 
 Grow first stream into second stream by connecting components.
-This allows to build more robust edge masks.
+This makes it possible to build more robust edge masks.
 
 This filter accepts the following options:
 
@@ -9663,7 +9666,7 @@ Scene change detection method. Scene change leads motion vectors to be in random
 @item none
 Disable scene change detection.
 @item fdiff
-Frame difference. Corresponding pixel values are compared and if it statisfies @var{scd_threshold} scene change is detected.
+Frame difference. Corresponding pixel values are compared and if it satisfies @var{scd_threshold} scene change is detected.
 @end table
 Default method is @samp{fdiff}.
 
@@ -16833,6 +16836,10 @@ Acceptable range is @code{[1, 7]}.
 Specify the bargraph gamma. Default value is @code{1}. Acceptable range is
 @code{[1, 7]}.
 
+@item bar_t
+Specify the bargraph transparency level. Lower value makes the bargraph sharper.
+Default value is @code{1}. Acceptable range is @code{[0, 1]}.
+
 @item timeclamp, tc
 Specify the transform timeclamp. At low frequency, there is trade-off between
 accuracy in time domain and frequency domain. If timeclamp is lower,
@@ -16936,6 +16943,11 @@ BT.2020 with non-constant luminance
 
 @end table
 
+@item cscheme
+Set spectrogram color scheme. This is list of floating point values with format
+@code{left_r|left_g|left_b|right_r|right_g|right_b}.
+The default is @code{1|0.5|0|0|0.5|1}.
+
 @end table
 
 @subsection Examples
@@ -17663,7 +17675,7 @@ magnitude across time and second represents phase across time.
 The filter will transform from frequency domain as displayed in videos back
 to time domain as presented in audio output.
 
-This filter is primarly created for reversing processed @ref{showspectrum}
+This filter is primarily created for reversing processed @ref{showspectrum}
 filter outputs, but can synthesize sound from other spectrograms too.
 But in such case results are going to be poor if the phase data is not
 available, because in such cases phase data need to be recreated, usually