Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On 7/9/15, Paul B Mahol one...@gmail.com wrote: From: LoRd_MuldeR muld...@gmx.de Signed-off-by: Paul B Mahol one...@gmail.com --- doc/filters.texi| 158 ++ libavfilter/Makefile| 1 + libavfilter/af_dynaudnorm.c | 739 libavfilter/allfilters.c| 1 + 4 files changed, 899 insertions(+) create mode 100644 libavfilter/af_dynaudnorm.c If nobody is against, I will apply this soon. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On 7/17/15, James Darnley james.darn...@gmail.com wrote:
On 2015-07-09 18:55, Paul B Mahol wrote:
+{ n, enable channels coupling, OFFSET(channels_coupled),
AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, FLAGS },
This should be enable channel coupling
Fixed
+static const double s_pi =
3.1415926535897932384626433832795028841971693993751058209749445923078164062862089986280348253421170679;
Why not use the standard M_PI define?
Its too short. And I try hard to produce exact same output as
reference implementation as possible, even if doubles are used.
+static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
+{
+double total_weight = 0.0;
+const double sigma = (((s-filter_size / 2.0) - 1.0) / 3.0) + (1.0 /
3.0);
+double adjust;
+int i;
+
+// Pre-computer constants
Pre-compute constants
Fixed.
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Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On 2015-07-09 18:55, Paul B Mahol wrote:
+{ n, enable channels coupling, OFFSET(channels_coupled),
AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, FLAGS },
This should be enable channel coupling
+static const double s_pi =
3.1415926535897932384626433832795028841971693993751058209749445923078164062862089986280348253421170679;
Why not use the standard M_PI define?
+static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
+{
+double total_weight = 0.0;
+const double sigma = (((s-filter_size / 2.0) - 1.0) / 3.0) + (1.0 /
3.0);
+double adjust;
+int i;
+
+// Pre-computer constants
Pre-compute constants
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Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On 2015-07-09 18:55, Paul B Mahol wrote:
diff --git a/doc/filters.texi b/doc/filters.texi
index 3fce874..74c408a 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1520,6 +1520,164 @@ Optional. It should have a value much less than 1
(e.g. 0.05 or 0.02) and is
used to prevent clipping.
@end table
+@section dynaudnorm
+Dynamic Audio Normalizer.
+
+This filter applies a certain amount of gain to the input audion in order
^^
audio
+to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
+contrast to more simple normalization algorithms, the Dynamic Audio
+Normalizer *dynamically* re-adjusts the gain factor to the input audio.
+This allows for applying extra gain to the quiet sections of the audio
+while avoiding distortions or clipping the loud sections. In other words:
+The Dynamic Audio Normalizer will even out the volume of quiet and loud
+sections, in the sense that the volume of each section is brought to the
+same target level. Note, however, that the Dynamic Audio Normalizer achieves
+this goal *without* applying dynamic range compressing. It will retain 100%
+of the dynamic range *within* each section of the audio file.
+
+@table @option
+@item 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,
+referred to as frames. This is required, because a peak magnitude has no
+meaning for just a single sample value. Instead, we need to determine the
+peak magnitude for a contiguous sequence of sample values. While a standard
+normalizer would simply use the peak magnitude of the complete file, the
+Dynamic Audio Normalizer determines the peak magnitude individually for each
+frame. The length of a frame is specified in milliseconds. By default, the
+Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has
+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.
+
+@item 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
+@code{window size} of the Gaussian smoothing filter. The filter's window size
+is specified in frames, centered around the current frame. For the sake of
+simplicity, this must be an odd number. Consequently, the default value of 31
+takes into account the current frame, as well as the 15 preceding frames and
+the 15 subsequent frames. Using a larger window results in a stronger
+smoothing effect and thus in less gain variation, i.e. slower gain
+adaptation. Conversely, using a smaller window results in a weaker smoothing
+effect and thus in more gain variation, i.e. faster gain adaptation.
+In other words, the more you increase this value, the more the Dynamic Audio
+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.
+
+@item 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
+makes sure that the normalized signal will never exceed the peak magnitude.
+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.
+
+@item 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
+result in clipping or distortion. The maximum gain factor is determined by
+the frame's highest magnitude sample. However, the Dynamic Audio Normalizer
+additionally bounds the frame's maximum gain factor by a predetermined
+(global) maximum gain factor. This is done in order to avoid excessive gain
+factors in silent or almost silent frames. By default, the maximum gain
+factor is 10.0, For most inputs the default value should be sufficient and
+it usually is not recommended to increase this value. Though, for input
+with an extremely low overall volume level, it may be necessary to allow even
+higher gain factors. Note, however, that the Dynamic Audio Normalizer does
+not simply apply a hard threshold (i.e. cut off values above the
threshold).
+Instead, a sigmoid threshold function will be applied. This way, the
+gainfactors will smoothly approach the threshold value, but never exceed that
^^^
Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On 7/17/15, James Darnley james.darn...@gmail.com wrote:
On 2015-07-09 18:55, Paul B Mahol wrote:
diff --git a/doc/filters.texi b/doc/filters.texi
index 3fce874..74c408a 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1520,6 +1520,164 @@ Optional. It should have a value much less than 1
(e.g. 0.05 or 0.02) and is
used to prevent clipping.
@end table
+@section dynaudnorm
+Dynamic Audio Normalizer.
+
+This filter applies a certain amount of gain to the input audion in
order
^^
audio
+to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
+contrast to more simple normalization algorithms, the Dynamic Audio
+Normalizer *dynamically* re-adjusts the gain factor to the input audio.
+This allows for applying extra gain to the quiet sections of the audio
+while avoiding distortions or clipping the loud sections. In other
words:
+The Dynamic Audio Normalizer will even out the volume of quiet and
loud
+sections, in the sense that the volume of each section is brought to the
+same target level. Note, however, that the Dynamic Audio Normalizer
achieves
+this goal *without* applying dynamic range compressing. It will retain
100%
+of the dynamic range *within* each section of the audio file.
+
+@table @option
+@item 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,
+referred to as frames. This is required, because a peak magnitude has no
+meaning for just a single sample value. Instead, we need to determine
the
+peak magnitude for a contiguous sequence of sample values. While a
standard
+normalizer would simply use the peak magnitude of the complete file, the
+Dynamic Audio Normalizer determines the peak magnitude individually for
each
+frame. The length of a frame is specified in milliseconds. By default,
the
+Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which
has
+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.
+
+@item 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
+@code{window size} of the Gaussian smoothing filter. The filter's window
size
+is specified in frames, centered around the current frame. For the sake
of
+simplicity, this must be an odd number. Consequently, the default value
of 31
+takes into account the current frame, as well as the 15 preceding frames
and
+the 15 subsequent frames. Using a larger window results in a stronger
+smoothing effect and thus in less gain variation, i.e. slower gain
+adaptation. Conversely, using a smaller window results in a weaker
smoothing
+effect and thus in more gain variation, i.e. faster gain adaptation.
+In other words, the more you increase this value, the more the Dynamic
Audio
+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.
+
+@item 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
+makes sure that the normalized signal will never exceed the peak
magnitude.
+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.
+
+@item 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
+result in clipping or distortion. The maximum gain factor is determined
by
+the frame's highest magnitude sample. However, the Dynamic Audio
Normalizer
+additionally bounds the frame's maximum gain factor by a predetermined
+(global) maximum gain factor. This is done in order to avoid excessive
gain
+factors in silent or almost silent frames. By default, the maximum
gain
+factor is 10.0, For most inputs the default value should be sufficient
and
+it usually is not recommended to increase this value. Though, for input
+with an extremely low overall volume level, it may be necessary to allow
even
+higher gain factors. Note, however, that the Dynamic Audio Normalizer
does
+not simply apply a hard threshold (i.e. cut off values above the
threshold).
+Instead, a sigmoid threshold function will be applied. This way, the
Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On 2015-07-17 12:55, Paul B Mahol wrote:
On 7/17/15, James Darnley james.darn...@gmail.com wrote:
On 2015-07-09 18:55, Paul B Mahol wrote:
+{ n, enable channels coupling, OFFSET(channels_coupled),
AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, FLAGS },
This should be enable channel coupling
Fixed
+static const double s_pi =
3.1415926535897932384626433832795028841971693993751058209749445923078164062862089986280348253421170679;
Why not use the standard M_PI define?
Its too short. And I try hard to produce exact same output as
reference implementation as possible, even if doubles are used.
I would have expected it to be long enough to be as precise as double
can be, or an exact hex define. Colour me surprised.
I don't object then.
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Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On Thu, 9 Jul 2015 16:55:48 + Paul B Mahol one...@gmail.com wrote: From: LoRd_MuldeR muld...@gmx.de Signed-off-by: Paul B Mahol one...@gmail.com --- doc/filters.texi| 158 ++ libavfilter/Makefile| 1 + libavfilter/af_dynaudnorm.c | 739 i think i prefer af dnormalize or so. we know its audio, so we dont need the aud part. make it a little easier to type and remember and search for. -compn ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH] avfilter: add Dynamic Audio Normalizer filter
On 7/10/15, compn te...@mi.rr.com wrote: On Thu, 9 Jul 2015 16:55:48 + Paul B Mahol one...@gmail.com wrote: From: LoRd_MuldeR muld...@gmx.de Signed-off-by: Paul B Mahol one...@gmail.com --- doc/filters.texi| 158 ++ libavfilter/Makefile| 1 + libavfilter/af_dynaudnorm.c | 739 i think i prefer af dnormalize or so. we know its audio, so we dont need the aud part. make it a little easier to type and remember and search for. This is how effect is called as sox plugin. And I would rather not rename it. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
