avfilter: add Dynamic Audio Normalizer filter

doc/filters.texi

@@ -1544,6 +1544,164 @@ 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 audio in order
+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
+gain factors will smoothly approach the threshold value, but never exceed that
+value.
+
+@item 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
+(only) by the frame's highest magnitude sample. This way, the samples can
+be amplified as much as possible without exceeding the maximum signal
+level, i.e. without clipping. Optionally, however, the Dynamic Audio
+Normalizer can also take into account the frame's root mean square,
+abbreviated RMS. In electrical engineering, the RMS is commonly used to
+determine the power of a time-varying signal. It is therefore considered
+that the RMS is a better approximation of the "perceived loudness" than
+just looking at the signal's peak magnitude. Consequently, by adjusting all
+frames to a constant RMS value, a uniform "perceived loudness" can be
+established. If a target RMS value has been specified, a frame's local gain
+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.
+
+@item 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.
+the maximum possible gain factor is determined by the "loudest" channel.
+However, in some recordings, it may happen that the volume of the different
+channels is uneven, e.g. one channel may be "quieter" than the other one(s).
+In this case, this option can be used to disable the channel coupling. This way,
+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.
+
+@item 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
+-1.0 to 1.0 range, regardless of the original input format. Normally, the
+audio signal, or "waveform", should be centered around the zero point.
+That means if we calculate the mean value of all samples in a file, or in a
+single frame, then the result should be 0.0 or at least very close to that
+value. If, however, there is a significant deviation of the mean value from
+0.0, in either positive or negative direction, this is referred to as a
+DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic
+Audio Normalizer provides optional DC bias correction.
+With DC bias correction enabled, the Dynamic Audio Normalizer will determine
+the mean value, or "DC correction" offset, of each input frame and subtract
+that value from all of the frame's sample values which ensures those samples
+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.
+
+@item 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
+subsequent frames. However, for the "boundary" frames, located at the very
+beginning and at the very end of the audio file, not all neighbouring
+frames are available. In particular, for the first few frames in the audio
+file, the preceding frames are not known. And, similarly, for the last few
+frames in the audio file, the subsequent frames are not known. Thus, the
+question arises which gain factors should be assumed for the missing frames
+in the "boundary" region. The Dynamic Audio Normalizer implements two modes
+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.
+
+@item 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
+full dynamic range will be retained within each local neighbourhood. However,
+in some cases it may be desirable to combine the Dynamic Audio Normalizer's
+normalization algorithm with a more "traditional" compression.
+For this purpose, the Dynamic Audio Normalizer provides an optional compression
+(thresholding) function. If (and only if) the compression feature is enabled,
+all input frames will be processed by a soft knee thresholding function prior
+to the actual normalization process. Put simply, the thresholding function is
+going to prune all samples whose magnitude exceeds a certain threshold value.
+However, the Dynamic Audio Normalizer does not simply apply a fixed threshold
+value. Instead, the threshold value will be adjusted for each individual
+frame.
+In general, smaller parameters result in stronger compression, and vice versa.
+Values below 3.0 are not recommended, because audible distortion may appear.
+@end table
+
 @section earwax
 
 Make audio easier to listen to on headphones.

libavfilter/Makefile

@@ -67,6 +67,7 @@ OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o
 OBJS-$(CONFIG_CHORUS_FILTER)                 += af_chorus.o generate_wave_table.o
 OBJS-$(CONFIG_COMPAND_FILTER)                += af_compand.o
 OBJS-$(CONFIG_DCSHIFT_FILTER)                += af_dcshift.o
+OBJS-$(CONFIG_DYNAUDNORM_FILTER)             += af_dynaudnorm.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
 OBJS-$(CONFIG_EBUR128_FILTER)                += f_ebur128.o
 OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o

libavfilter/allfilters.c

@@ -83,6 +83,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(CHORUS,         chorus,         af);
     REGISTER_FILTER(COMPAND,        compand,        af);
     REGISTER_FILTER(DCSHIFT,        dcshift,        af);
+    REGISTER_FILTER(DYNAUDNORM,     dynaudnorm,     af);
     REGISTER_FILTER(EARWAX,         earwax,         af);
     REGISTER_FILTER(EBUR128,        ebur128,        af);
     REGISTER_FILTER(EQUALIZER,      equalizer,      af);