[FFmpeg-cvslog] avfilter/af_dynaudnorm: add more descriptive aliases for options
Paul B Mahol
git at videolan.org
Wed Jul 24 17:24:06 EEST 2019
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Jul 24 14:16:49 2019 +0200| [73afea3cccbd1f6a8bdb834ace7984440ee783f7] | committer: Kieran Kunhya
avfilter/af_dynaudnorm: add more descriptive aliases for options
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=73afea3cccbd1f6a8bdb834ace7984440ee783f7
---
doc/filters.texi | 18 +++++++++---------
libavfilter/af_dynaudnorm.c | 27 ++++++++++++++++++---------
2 files changed, 27 insertions(+), 18 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index b48a0cda91..c4ba907981 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3135,7 +3135,7 @@ this goal *without* applying "dynamic range compressing". It will retain 100%
of the dynamic range *within* each section of the audio file.
@table @option
- at item f
+ at item framelen, f
Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.
Default is 500 milliseconds.
The Dynamic Audio Normalizer processes the input audio in small chunks,
@@ -3150,7 +3150,7 @@ been found to give good results with most files.
Note that the exact frame length, in number of samples, will be determined
automatically, based on the sampling rate of the individual input audio file.
- at item g
+ at item gausssize, g
Set the Gaussian filter window size. In range from 3 to 301, must be odd
number. Default is 31.
Probably the most important parameter of the Dynamic Audio Normalizer is the
@@ -3167,7 +3167,7 @@ Normalizer will behave like a "traditional" normalization filter. On the
contrary, the more you decrease this value, the more the Dynamic Audio
Normalizer will behave like a dynamic range compressor.
- at item p
+ at item peak, p
Set the target peak value. This specifies the highest permissible magnitude
level for the normalized audio input. This filter will try to approach the
target peak magnitude as closely as possible, but at the same time it also
@@ -3176,7 +3176,7 @@ A frame's maximum local gain factor is imposed directly by the target peak
magnitude. The default value is 0.95 and thus leaves a headroom of 5%*.
It is not recommended to go above this value.
- at item m
+ at item maxgain, m
Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.
The Dynamic Audio Normalizer determines the maximum possible (local) gain
factor for each input frame, i.e. the maximum gain factor that does not
@@ -3194,7 +3194,7 @@ Instead, a "sigmoid" threshold function will be applied. This way, the
gain factors will smoothly approach the threshold value, but never exceed that
value.
- at item r
+ at item targetrms, r
Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.
By default, the Dynamic Audio Normalizer performs "peak" normalization.
This means that the maximum local gain factor for each frame is defined
@@ -3212,7 +3212,7 @@ factor is defined as the factor that would result in exactly that RMS value.
Note, however, that the maximum local gain factor is still restricted by the
frame's highest magnitude sample, in order to prevent clipping.
- at item n
+ at item coupling, n
Enable channels coupling. By default is enabled.
By default, the Dynamic Audio Normalizer will amplify all channels by the same
amount. This means the same gain factor will be applied to all channels, i.e.
@@ -3224,7 +3224,7 @@ the gain factor will be determined independently for each channel, depending
only on the individual channel's highest magnitude sample. This allows for
harmonizing the volume of the different channels.
- at item c
+ at item correctdc, c
Enable DC bias correction. By default is disabled.
An audio signal (in the time domain) is a sequence of sample values.
In the Dynamic Audio Normalizer these sample values are represented in the
@@ -3243,7 +3243,7 @@ are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
boundaries, the DC correction offset values will be interpolated smoothly
between neighbouring frames.
- at item b
+ at item altboundary, b
Enable alternative boundary mode. By default is disabled.
The Dynamic Audio Normalizer takes into account a certain neighbourhood
around each frame. This includes the preceding frames as well as the
@@ -3258,7 +3258,7 @@ to deal with this situation. The default boundary mode assumes a gain factor
of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and
"fade out" at the beginning and at the end of the input, respectively.
- at item s
+ at item compress, s
Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
By default, the Dynamic Audio Normalizer does not apply "traditional"
compression. This means that signal peaks will not be pruned and thus the
diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c
index fd430884d7..639503384e 100644
--- a/libavfilter/af_dynaudnorm.c
+++ b/libavfilter/af_dynaudnorm.c
@@ -82,15 +82,24 @@ typedef struct DynamicAudioNormalizerContext {
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption dynaudnorm_options[] = {
- { "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
- { "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
- { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
- { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
- { "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
- { "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
- { "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
- { "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
- { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
+ { "framelen", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
+ { "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
+ { "gausssize", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
+ { "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
+ { "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
+ { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
+ { "maxgain", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
+ { "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
+ { "targetrms", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
+ { "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
+ { "coupling", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
+ { "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
+ { "correctdc", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
+ { "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
+ { "altboundary", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
+ { "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
+ { "compress", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
+ { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
{ NULL }
};
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