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Can be upward or downward. Default is downward. If a signal of stream rises above this level it will affect the gain reduction. By default it is 0. Range is between 0. Set a ratio by which the signal is reduced. Default is 2. Range is between 1 and Amount of milliseconds the signal has to rise above the threshold before gain reduction starts. Default is Amount of milliseconds the signal has to fall below the threshold before reduction is decreased again. Set the amount by how much signal will be amplified after processing.

Default is 1. Range is from 1 to Curve the sharp knee around the threshold to enter gain reduction more softly. Range is between 1 and 8. Choose if the average level between all channels of input stream or the louder maximum channel of input stream affects the reduction.

Default is average. Should the exact signal be taken in case of peak or an RMS one in case of rms. Default is rms which is mostly smoother. Apply cross fade from one input audio stream to another input audio stream. The cross fade is applied for specified duration near the end of first stream.

Specify the number of samples for which the cross fade effect has to last. At the end of the cross fade effect the first input audio will be completely silent. Specify the duration of the cross fade effect. See ffmpeg-utils the Time duration section in the ffmpeg-utils 1 manual for the accepted syntax.

This filter splits audio stream into two or more frequency ranges. Summing all streams back will give flat output. Set filter order for each band split. This controls filter roll-off or steepness of filter transfer function. Available values are:. 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.

Material reduced in bit depth sounds more harsh and "digital". This filter is able to even round to continuous values instead of discrete bit depths. An Anti-Aliasing setting is able to produce "softer" crushing sounds. Another feature of this filter is the logarithmic mode. This setting switches from linear distances between bits to logarithmic ones.

The human ear has a logarithmic perception, so this kind of crushing is much more pleasant. Logarithmic crushing is also able to get anti-aliased. Samples detected as impulsive noise are replaced by interpolated samples using autoregressive modelling. Set window size, in milliseconds. Allowed range is from 10 to Default value is 55 milliseconds.

This sets size of window which will be processed at once. Set window overlap, in percentage of window size. Allowed range is from 50 to Default value is 75 percent. Setting this to a very high value increases impulsive noise removal but makes whole process much slower. Set autoregression order, in percentage of window size. Allowed range is from 0 to Default value is 2 percent. This option also controls quality of interpolated samples using neighbour good samples.

Set threshold value. Allowed range is from 1 to Default value is 2. This controls the strength of impulsive noise which is going to be removed. The lower value, the more samples will be detected as impulsive noise. Set burst fusion, in percentage of window size. Allowed range is 0 to If any two samples detected as noise are spaced less than this value then any sample between those two samples will be also detected as noise. Default value is 8 percent.

Default value is Higher values make clip detection less aggressive. Set size of histogram used to detect clips. Allowed range is from to Unused delays will be silently ignored. If number of given delays is smaller than number of channels all remaining channels will not be delayed. Use last set delay for all remaining channels. By default is disabled. This option if enabled changes how option delays is interpreted.

Echoes are reflected sound and can occur naturally amongst mountains and sometimes large buildings when talking or shouting; digital echo effects emulate this behaviour and are often used to help fill out the sound of a single instrument or vocal. The time difference between the original signal and the reflection is the delay , and the loudness of the reflected signal is the decay.

Multiple echoes can have different delays and decays. Allowed range for each delay is 0 - Allowed range for each decay is 0 - 1. Default is 0. Audio emphasis filter creates or restores material directly taken from LPs or emphased CDs with different filter curves.

Once the material is played back the inverse filter has to be applied to restore the distortion of the frequency response. Set filter mode. For restoring material use reproduction mode, otherwise use production mode. Default is reproduction mode. This filter accepts one or more expressions one for each channel , which are evaluated and used to modify a corresponding audio signal. If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.

Set output channel layout. If not specified, the channel layout is specified by the number of expressions. An exciter is used to produce high sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. An Exciter raises the upper end of an audio signal without simply raising the higher frequencies like an equalizer would do to create a more "crisp" or "brilliant" sound.

Set the amount of harmonics added to original signal. Set the amount of newly created harmonics. Allowed range is from 0. Default value is 8. Set the lower frequency limit of producing harmonics in Hz. Allowed range is from to Hz. Default is Hz. Set the upper frequency limit of producing harmonics.

If value is lower than Hz no limit is applied. Specify the effect type, can be either in for fade-in, or out for a fade-out effect. Default is in. Specify the number of samples for which the fade effect has to last. At the end of the fade-in effect the output audio will have the same volume as the input audio, at the end of the fade-out transition the output audio will be silence.

Specify the start time of the fade effect. The value must be specified as a time duration; see ffmpeg-utils the Time duration section in the ffmpeg-utils 1 manual for the accepted syntax. Specify the duration of the fade effect. Set custom band noise for every one of 15 bands. Enable noise tracking. With this enabled, noise floor is automatically adjusted. Start or stop measuring noise profile. Syntax for the command is : "start" or "stop" string. After measuring noise profile is stopped it will be automatically applied in filtering.

Change noise reduction. Argument is single float number. Change noise floor. Change output mode operation. Syntax for the command is : "i", "o" or "n" string. Default is "re". Default is "im". Each expression in real and imag can contain the following constants and functions:. Return the value of imaginary part of frequency bin at location bin , channel. Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. It can be used as component for digital crossover filters, room equalization, cross talk cancellation, wavefield synthesis, auralization, ambiophonics, ambisonics and spatialization.

This filter uses the streams higher than first one as FIR coefficients. If the non-first stream holds a single channel, it will be used for all input channels in the first stream, otherwise the number of channels in the non-first stream must be same as the number of channels in the first stream.

Set gain to be applied to IR coefficients before filtering. Allowed range is 0 to 1. This gain is applied after any gain applied with gtype option. Set format of IR stream. Can be mono or input. Default is input.

Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds. Allowed range is 0. Show IR frequency response, magnitude magenta , phase green and group delay yellow in additional video stream. By default it is disabled. Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled. Set minimal partition size used for convolution. Lower values decreases latency at cost of higher CPU usage.

Set maximal partition size used for convolution. Allowed range is from 8 to Lower values may increase CPU usage. Set number of input impulse responses streams which will be switchable at runtime. Set IR stream which will be used for convolution, starting from 0 , should always be lower than supplied value by nbirs option. This option can be changed at runtime via commands. Set output format constraints for the input audio. The framework will negotiate the most appropriate format to minimize conversions.

See ffmpeg-utils the Channel Layout section in the ffmpeg-utils 1 manual for the required syntax. Set the noise sigma, allowed range is from 0 to 1. This option controls strength of denoising applied to input samples. Most useful way to set this option is via decibels, eg. Set the number of wavelet levels of decomposition. Setting this too low make denoising performance very poor.

Set wavelet type for decomposition of input frame. They are sorted by number of coefficients, from lowest to highest. More coefficients means worse filtering speed, but overall better quality. Available wavelets are:. Set percent of full denoising. Allowed range is from 0 to percent. Default value is 85 percent or partial denoising. If enabled, first input frame will be used as noise profile. If first frame samples contain non-noise performance will be very poor.

If enabled, input frames are analyzed for presence of noise. If noise is detected with high possibility then input frame profile will be used for processing following frames, until new noise frame is detected. Set size of single frame in number of samples. Default frame size is samples. Set softness applied inside thresholding function. Default softness is 1. A gate is mainly used to reduce lower parts of a signal.

This kind of signal processing reduces disturbing noise between useful signals. Gating is done by detecting the volume below a chosen level threshold and dividing it by the factor set with ratio. The bottom of the noise floor is set via range. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over time.

This is done by setting attack and release. Set the mode of operation. If set to upward mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of downward lower parts of signal will be reduced. Set the level of gain reduction when the signal is below the threshold.

Allowed range is from 0 to 1. Setting this to 0 disables reduction and then filter behaves like expander. If a signal rises above this level the gain reduction is released. Amount of milliseconds the signal has to rise above the threshold before gain reduction stops. Default is 20 milliseconds. Amount of milliseconds the signal has to fall below the threshold before the reduction is increased again. Default is milliseconds. Set amount of amplification of signal after processing. Allowed range is from 1 to 8.

Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms. Choose if the average level between all channels or the louder channel affects the reduction. Can be average or maximum. Normalize filter coefficients, by default is enabled. Enabling it will normalize magnitude response at DC to 0dB.

Coefficients in tf and sf format are separated by spaces and are in ascending order. Coefficients in zp format are complex numbers with i imaginary unit. Last provided coefficients will be used for all remaining channels. The limiter prevents an input signal from rising over a desired threshold. This limiter uses lookahead technology to prevent your signal from distorting. It means that there is a small delay after the signal is processed.

Keep in mind that the delay it produces is the attack time you set. The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5 milliseconds. Come back from limiting to attenuation 1. Default is 50 milliseconds. When gain reduction is always needed ASC takes care of releasing to an average reduction level rather than reaching a reduction of 0 in the release time.

Select how much the release time is affected by ASC, 0 means nearly no changes in release time while 1 produces higher release times. Depending on picked setting it is recommended to upsample input 2x or 4x times with aresample before applying this filter. Apply a two-pole all-pass filter with central frequency in Hz frequency , and filter-width width.

Normalize biquad coefficients, by default is disabled. If the channel layouts of the inputs are disjoint, and therefore compatible, the channel layout of the output will be set accordingly and the channels will be reordered as necessary.

If the channel layouts of the inputs are not disjoint, the output will have all the channels of the first input then all the channels of the second input, in that order, and the channel layout of the output will be the default value corresponding to the total number of channels. For example, if the first input is in 2.

On the other hand, if both input are in stereo, the output channels will be in the default order: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4. Note that this filter only supports float samples the amerge and pan audio filters support many formats. If the amix input has integer samples then aresample will be automatically inserted to perform the conversion to float samples.

The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds. Specify weight of each input audio stream as sequence. Each weight is separated by space. By default all inputs have same weight. Always scale inputs instead of only doing summation of samples.

Beware of heavy clipping if inputs are not normalized prior or after filtering by this filter if this option is disabled. By default is enabled. Multiply first audio stream with second audio stream and store result in output audio stream. Multiplication is done by multiplying each sample from first stream with sample at same position from second stream. Set channel number to which equalization will be applied.

Set max gain that will be displayed. Only useful if curves option is activated. 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. Set frequency scale used to draw frequency response in video output. Can be linear or logarithmic.

Default is logarithmic. Set color for each channel curve which is going to be displayed in video stream. Unrecognised or missing colors will be replaced by white color. Alter existing filter parameters. Each sample is adjusted by looking for other samples with similar contexts. This context similarity is defined by comparing their surrounding patches of size p.

Patches are searched in an area of r around the sample. Set patch radius duration. Allowed range is from 1 to milliseconds. Default value is 2 milliseconds. Set research radius duration. Allowed range is from 2 to milliseconds. Default value is 6 milliseconds. Set smooth factor. Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream. This adaptive filter is used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean square of the error signal difference between the desired, 2nd input audio stream and the actual signal, the 1st input audio stream.

This can be used together with ffmpeg -shortest to extend audio streams to the same length as the video stream. Set the number of samples of silence to add to the end. After the value is reached, the stream is terminated. Set the minimum total number of samples in the output audio stream. If the value is longer than the input audio length, silence is added to the end, until the value is reached. Specify the duration of samples of silence to add. Used only if set to non-zero value.

Specify the minimum total duration in the output audio stream. A phaser filter creates series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect. Set number of iterations of psychoacoustic clipper. Audio pulsator is something between an autopanner and a tremolo.

But it can produce funny stereo effects as well. Pulsator changes the volume of the left and right channel based on a LFO low frequency oscillator with different waveforms and shifted phases. This filter have the ability to define an offset between left and right channel.

An offset of 0 means that both LFO shapes match each other. The left and right channel are altered equally - a conventional tremolo. At 1 both curves match again. Every setting in between moves the phase shift gapless between all stages and produces some "bypassing" sounds with sine and triangle waveforms. The more you set the offset near 1 starting from the 0. Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or sawdown.

Default is sine. Set frequency in Hz. Allowed range is [0. Only used if timing is set to hz. Resample the input audio to the specified parameters, using the libswresample library. If none are specified then the filter will automatically convert between its input and output.

See the ffmpeg-resampler "Resampler Options" section in the ffmpeg-resampler 1 manual for the complete list of supported options. Set how much to mix filtered samples into final output. Allowed range is from -1 to 1.

Negative values are special, they set how much to keep filtered noise in the final filter output. Set this option to -1 to hear actual noise removed from input signal. This filter takes two audio streams for input, and outputs first audio stream.

Results are in dB per channel at end of either input. The last output packet may contain a different number of samples, as the filter will flush all the remaining samples when the input audio signals its end. Set the number of frames per each output audio frame. The number is intended as the number of samples per each channel. If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame will contain the same number of samples as the previous ones.

For example, to set the number of per-frame samples to and disable padding for the last frame, use:. Set the sample rate without altering the PCM data. This will result in a change of speed and pitch. Show a line containing various information for each input audio frame. The input audio is not modified. The Adler checksum printed in hexadecimal of the audio data. For planar audio, the data is treated as if all the planes were concatenated. Soft clipping is a type of distortion effect where the amplitude of a signal is saturated along a smooth curve, rather than the abrupt shape of hard-clipping.

This filter uses PocketSphinx for speech recognition. To enable compilation of this filter, you need to configure FFmpeg with --enable-pocketsphinx. Set sampling rate of input audio. Defaults is This need to match speech models, otherwise one will get poor results. Display time domain statistical information about the audio channels.

Statistics are calculated and displayed for each audio channel and, where applicable, an overall figure is also given. Short window length in seconds, used for peak and trough RMS measurement. Allowed range is [0 - 10]. Set metadata injection.

All the metadata keys are prefixed with lavfi. X , where X is channel number starting from 1 or string Overall. Default is disabled. For example full key look like this lavfi. Select the entries which need to be measured per channel.

The metadata keys can be used as flags, default is all which measures everything. Select the entries which need to be measured overall. Mean difference between two consecutive samples. The average of each difference between two consecutive samples. Flatness i. Number of occasions not the number of samples that the signal attained either Min level or Max level. Entropy measured across whole audio. Entropy of value near 1.

Set dry gain, how much of original signal is kept. Set wet gain, how much of filtered signal is kept. This filter allows to set custom, steeper roll off than highpass filter, and thus is able to more attenuate frequency content in stop-band.

The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume nominal 1. Tempo must be in the [0. Note that tempo greater than 2 will skip some samples rather than blend them in. If for any reason this is a concern it is always possible to daisy-chain several instances of atempo to achieve the desired product tempo.

Timestamp in seconds of the start of the section to keep. Specify time of the first audio sample that will be dropped, i. Same as start , except this option sets the start timestamp in samples instead of seconds. Also note that this filter does not modify the timestamps.

If you wish to have the output timestamps start at zero, insert the asetpts filter after the atrim filter. If multiple start or end options are set, this filter tries to be greedy and keep all samples that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple atrim filters. The defaults are such that all the input is kept.

So it is possible to set e. Resulted samples are always between -1 and 1 inclusive. If result is 1 it means two input samples are highly correlated in that selected segment. Result 0 means they are not correlated at all. If result is -1 it means two input samples are out of phase, which means they cancel each other. Set size of segment over which cross-correlation is calculated.

Allowed range is from 2 to Set algorithm for cross-correlation. Can be slow or fast. Default is slow. Fast algorithm assumes mean values over any given segment are always zero and thus need much less calculations to make. This is generally not true, but is valid for typical audio streams. Apply a two-pole Butterworth band-pass filter with central frequency frequency , and 3dB-point band-width width.

The filter roll off at 6dB per octave 20dB per decade. Apply a two-pole Butterworth band-reject filter with central frequency frequency , and 3dB-point band-width width. This is also known as shelving equalisation EQ.

Give the gain at 0 Hz. Beware of clipping when using a positive gain. The default value is Hz. Apply a biquad IIR filter with the given coefficients. Where b0 , b1 , b2 and a0 , a1 , a2 are the numerator and denominator coefficients respectively.

Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records. To enable compilation of this filter you need to configure FFmpeg with --enable-libbs2b. Map channels from input to output. FL for front left or its index in the input channel layout. If no mapping is present, the filter will implicitly map input channels to output channels, preserving indices.

A channel layout describing the channels to be extracted as separate output streams or "all" to extract each input channel as a separate stream. The default is "all". Chorus resembles an echo effect with a short delay, but whereas with echo the delay is constant, with chorus, it is varied using using sinusoidal or triangular modulation.

The modulation depth defines the range the modulated delay is played before or after the delay. Hence the delayed sound will sound slower or faster, that is the delayed sound tuned around the original one, like in a chorus where some vocals are slightly off key.

A list of times in seconds for each channel over which the instantaneous level of the input signal is averaged to determine its volume. For most situations, the attack time response to the audio getting louder should be shorter than the decay time, because the human ear is more sensitive to sudden loud audio than sudden soft audio.

A typical value for attack is 0. A list of points for the transfer function, specified in dB relative to the maximum possible signal amplitude. The input values must be in strictly increasing order but the transfer function does not have to be monotonically rising. Set the additional gain in dB to be applied at all points on the transfer function. This allows for easy adjustment of the overall gain. It defaults to 0. Set an initial volume, in dB, to be assumed for each channel when filtering starts.

This permits the user to supply a nominal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding has begun to operate. A typical value for audio which is initially quiet is dB. Set a delay, in seconds.

The input audio is analyzed immediately, but audio is delayed before being fed to the volume adjuster. Compensation Delay Line is a metric based delay to compensate differing positions of microphones or speakers. For example, you have recorded guitar with two microphones placed in different locations.

Because the front of sound wave has fixed speed in normal conditions, the phasing of microphones can vary and depends on their location and interposition. The best sound mix can be achieved when these microphones are in phase synchronized.

That makes the final mix sound moody. This filter helps to solve phasing problems by adding different delays to each microphone track and make them synchronized. The best result can be reached when you take one track as base and synchronize other tracks one by one with it. Higher sample rates will give more tolerance. Crossfeed is the process of blending the left and right channels of stereo audio recording. It is mainly used to reduce extreme stereo separation of low frequencies.

Set strength of crossfeed. This sets gain of low shelf filter for side part of stereo image. Default is -6dB. Max allowed is db when strength is set to 1. Set soundstage wideness. This sets cut off frequency of low shelf filter. Default is cut off near Hz. With range set to 1 cut off frequency is set to Hz. Sets the intensity of effect default: 2.

Must be in range between To inverse filtering use negative value. This can be useful to remove a DC offset caused perhaps by a hardware problem in the recording chain from the audio. The effect of a DC offset is reduced headroom and hence volume. The astats filter can be used to determine if a signal has a DC offset. It should have a value much less than 1 e.

How much of original frequency content to keep when de-essing. DR values of 14 and higher is found in very dynamic material. DR of 8 to 13 is found in transition material. And anything less that 8 have very poor dynamics and is very compressed. Set window length in seconds used to split audio into segments of equal length. Default is 3 seconds.

This filter applies a certain amount of gain to the input audio in order to bring its peak magnitude to a target level e. 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.

Set the frame length in milliseconds. In range from 10 to 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 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. Set the Gaussian filter window size. In range from 3 to , must be odd number. Probably the most important parameter of the Dynamic Audio Normalizer is the window size of the Gaussian smoothing filter. 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. Conversely, using a smaller window results in a weaker smoothing effect and thus in more gain variation, i. 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.

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. The default value is 0. It is not recommended to go above this value.

Set the maximum gain factor. In range from 1. The Dynamic Audio Normalizer determines the maximum possible local gain factor for each input frame, i. This is done in order to avoid excessive gain factors in "silent" or almost silent frames. By default, the maximum gain factor is 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. Instead, a "sigmoid" threshold function will be applied. This way, the gain factors will smoothly approach the threshold value, but never exceed that value. Set the target RMS. In range from 0. By default, the Dynamic Audio Normalizer performs "peak" normalization.

This way, the samples can be amplified as much as possible without exceeding the maximum signal level, i. In electrical engineering, the RMS is commonly used to determine the power of a time-varying signal. Consequently, by adjusting all frames to a constant RMS value, a uniform "perceived loudness" can be established. Enable channels coupling. 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. However, in some recordings, it may happen that the volume of the different channels is uneven, e. In this case, this option can be used to disable the channel coupling.

This allows for harmonizing the volume of the different channels. Enable DC bias correction. 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 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. If, however, there is a significant deviation of the mean value from 0.

Also, in order to avoid "gaps" at the frame boundaries, the DC correction offset values will be interpolated smoothly between neighbouring frames. Enable alternative boundary mode. 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. Set the compress factor. 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. 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. Set the target threshold value. This specifies the lowest permissible magnitude level for the audio input which will be normalized. If input frame volume is above this value frame will be normalized. Otherwise frame may not be normalized at all.

The default value is set to 0, which means all input frames will be normalized. This option is mostly useful if digital noise is not wanted to be amplified. Apply a two-pole peaking equalisation EQ filter. With this filter, the signal-level at and around a selected frequency can be increased or decreased, whilst unlike bandpass and bandreject filters that at all other frequencies is unchanged.

In order to produce complex equalisation curves, this filter can be given several times, each with a different central frequency. Linearly increases the difference between left and right channels which adds some sort of "live" effect to playback. Sets the difference coefficient default: 2. If enabled, use fixed number of audio samples. This improves speed when filtering with large delay. Enable 2-channel convolution using complex FFT.

This improves speed significantly. Set swept wave shape, can be triangular or sinusoidal. Not all file systems follow the tilde substitution convention, and systems can be configured to disable 8. Therefore, do not make the assumption that the 8.

To request 8. This is true even if a long file name contains extended characters, regardless of the code page that is active during a disk read or write operation. Files using long file names can be copied between NTFS file system partitions and Windows FAT file system partitions without losing any file name information.

In this case, the short file name is substituted if possible. The path to a specified file consists of one or more components , separated by a special character a backslash , with each component usually being a directory name or file name, but with some notable exceptions discussed below. This prefix determines the namespace the path is using, and additionally what special characters are used in which position within the path, including the last character.

Each component of a path will also be constrained by the maximum length specified for a particular file system. In general, these rules fall into two categories: short and long. Note that directory names are stored by the file system as a special type of file, but naming rules for files also apply to directory names. To summarize, a path is simply the string representation of the hierarchy between all of the directories that exist for a particular file or directory name.

For Windows API functions that manipulate files, file names can often be relative to the current directory, while some APIs require a fully qualified path. A file name is relative to the current directory if it does not begin with one of the following:. If a file name begins with only a disk designator but not the backslash after the colon, it is interpreted as a relative path to the current directory on the drive with the specified letter.

Note that the current directory may or may not be the root directory depending on what it was set to during the most recent "change directory" operation on that disk. Examples of this format are as follows:. A path is also said to be relative if it contains "double-dots"; that is, two periods together in one component of the path.

This special specifier is used to denote the directory above the current directory, otherwise known as the "parent directory". Relative paths can combine both example types, for example "C This is useful because, although the system keeps track of the current drive along with the current directory of that drive, it also keeps track of the current directories in each of the different drive letters if your system has more than one , regardless of which drive designator is set as the current drive.

In later versions of Windows, changing a registry key or using the Group Policy tool is required to remove the limit. See Maximum Path Length Limitation for full details. There are two main categories of namespace conventions used in the Windows APIs, commonly referred to as NT namespaces and the Win32 namespaces. The NT namespace was designed to be the lowest level namespace on which other subsystems and namespaces could exist, including the Win32 subsystem and, by extension, the Win32 namespaces.

Early versions of Windows also defined several predefined, or reserved, names for certain special devices such as communications serial and parallel ports and the default display console as part of what is now called the NT device namespace, and are still supported in current versions of Windows for backward compatibility.

The Win32 namespace prefixing and conventions are summarized in this section and the following section, with descriptions of how they are used. Note that these examples are intended for use with the Windows API functions and do not all necessarily work with Windows shell applications such as Windows Explorer. For this reason there is a wider range of possible paths than is usually available from Windows shell applications, and Windows applications that take advantage of this can be developed using these namespace conventions.

For more information about the normal maximum path limitation, see the previous section Maximum Path Length Limitation. This is how access to physical disks and volumes is accomplished directly, without going through the file system, if the API supports this type of access. You can access many devices other than disks this way using the CreateFile and DefineDosDevice functions, for example. This allows you to access those devices directly, bypassing the file system.

This works because these device names are created by the system as these devices are enumerated, and some drivers will also create other aliases in the system. Always check the reference topic for each API to be sure. To illustrate, it is useful to browse the Windows namespaces in the system object browser using the Windows Sysinternals WinObj tool. The subfolder called "Global?? Named device objects reside in the NT namespace within the "Device" subdirectory.

Here you may also find Serial0 and Serial1, the device objects representing the first two COM ports if present on your system. A device object representing a volume would be something like "HarddiskVolume1", although the numeric suffix may vary. The name "DR0" under subdirectory "Harddisk0" is an example of the device object representing a disk, and so on. To make these device objects accessible by Windows applications, the device drivers create a symbolic link symlink in the Win32 namespace, "Global??

Without a symlink, a specified device "Xxx" will not be available to any Windows application using Win32 namespace conventions as described previously. With the addition of multi-user support via Terminal Services and virtual machines, it has further become necessary to virtualize the system-wide root device within the Win32 namespace. This prefix ensures that the path following it looks in the true root path of the system object manager and not a session-dependent path.

File System Functionality Comparison.

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