The first and second mono files would become the left and right channels of the stereo file. If the number of channels in the input files is not the same, the missing channels are considered to contain all zero. When combining input files, SoX applies any specified effects including, for example, the vol volume adjustment effect after the audio has been combined. However, it is often useful to be able to set the volume of i. If it is given for only some of the input files then the others receive no volume adjustment.

In some circumstances, automatic volume adjustments may be applied see below. There are some special considerations that need to made when mixing input files:. If this results in audio that is too quiet or otherwise unbalanced then the input file volumes can be set manually as described above.

Using the norm effect on the mix is another alternative. If mixed audio seems loud enough at some points but too quiet in others then dynamic range compression should be applied to correct this – see the compand effect. Note that this balancing factor does not guarantee that clipping will not occur, but the number of clips will usually be low and the resultant distortion is generally imperceptible.

SoX will then enter multiple output mode. In multiple output mode, a unique number will automatically be appended to the end of all filenames. If the filename has an extension then the number is inserted before the extension.

If end of file is reached before the effects chain stops itself then no new file will be created as it would be empty. The following is an example of splitting the first 60 seconds of an input file into two 30 second files and ignoring the rest.

Stopping SoX Usually SoX will complete its processing and exit automatically once it has read all available audio data from the input files. If desired, it can be terminated earlier by sending an interrupt signal to the process usually by pressing the keyboard interrupt key which is normally Ctrl-C.

This is a natural requirement in some circumstances, e. Note that when using SoX to play multiple files, Ctrl-C behaves slightly differently: pressing it once causes SoX to skip to the next file; pressing it twice in quick succession causes SoX to exit.

Another option to stop processing early is to use an effect that has a time period or sample count to determine the stopping point.

The trim effect is an example of this. Once all effects chains have stopped then SoX will also stop. Filenames can be simple file names, absolute or relative path names, or URLs input files only. Note that URL support requires that wget 1 is available. Note: Giving SoX an input or output filename that is the same as a SoX effect-name will not work since SoX will treat it as an effect specification.

The only work-around to this is to avoid such filenames. Special Filenames The following special filenames may be used in certain circumstances in place of a normal filename on the command line:. This allows a single set of file options to be applied to a group of files.

This can be used in place of an output filename to specify that the SoX command should be used as in input pipe to another SoX command. For example, the command:. This can be used in place of an input or output filename to specify that the default audio device if one has been built into SoX is to be used. This is akin to invoking rec or play as described above.

Using a null file to input audio is equivalent to using a normal audio file that contains an infinite amount of silence, and as such is not generally useful unless used with an effect that specifies a finite time length such as trim or synth. Using a null file to output audio amounts to discarding the audio and is useful mainly with effects that produce information about the audio instead of affecting it such as noiseprof or stat. The sampling rate associated with a null file is by default 48 kHz, but, as with a normal file, this can be overridden if desired using command-line format options see below.

Global Options These options can be specified on the command line at any point before the first effect name. For example:. An alternative approach is to explicitly invoke SoX with default option values, e.

Note that the way to set environment variables varies from system to system. Here are some examples:. Set the size in bytes of the buffers used for processing audio default This is the default behaviour. See Input File Combining above for a description of the different combining methods.

An example of why this might occasionally be useful is if a file has been converted from 16 to 24 bit with the intention of doing some processing on it, but in fact no processing is needed after all and the original 16 bit file has been lost, then, strictly speaking, no dither is needed if converting the file back to 16 bit.

See also the stats effect for how to determine the actual bit depth of the audio within a file. The file is parsed as if the values were specified on the command line. A new line can be used in place of the special : marker to separate effect chains. For convenience, such markers at the end of the file are normally ignored; if you want to specify an empty last effects chain, use an explicit : by itself on the last line of the file.

This option causes any effects specified on the command line to be discarded. Automatically invoke the gain effect to guard against clipping. Show usage information on the specified effect. The name all can be used to show usage on all effects. Show information about the specified file format.

The name all can be used to show information on all formats. Only if given as the first parameter to sox , behave as soxi 1. This may reduce processing time, though sometimes it may be necessary to use this option in conjunction with a larger buffer size than is the default to gain any benefit from multi-threaded processing e. Prompt before overwriting an existing file with the same name as that given for the output file.

Unintentionally overwriting a file is easier than you might think, for example, if you accidentally enter. Hence, using this option is recommended. Automatically invoke the gain effect to guard against clipping and to normalise the audio. Optionally, the audio can be normalized to a given level usually below 0 dBFS:.

When this option is given, where applicable, SoX will embed a fixed time-stamp in the output file e. Select whether or not to apply replay-gain adjustment to input files. The default is off for sox and rec , album for play where at least the first two input files are tagged with the same Artist and Album names, and track for play otherwise.

Also shown is a peak-level meter, and an indication if clipping has occurred. The peak-level meter shows up to two channels and is calibrated for digital audio as follows right channel shown :.

A three-second peak-held value of headroom in dBs will be shown to the right of the meter if this is below 6dB. This option is enabled by default when using SoX to play or record audio. This can be useful if there are permission or free-space problems with the default location.

Set verbosity. This is particularly useful for seeing how any automatic effects have been invoked by SoX. SoX displays messages on the console stderr according to the following verbosity levels:. Only error messages are shown. These are generated if SoX cannot complete the requested commands. Warning messages are also shown. These are generated if SoX can complete the requested commands, but not exactly according to the requested command parameters, or if clipping occurs.

Useful for seeing exactly how SoX is processing your audio. By default, the verbosity level is set to 2 shows errors and warnings. Input File Options These options apply only to input files and may precede only input filenames on the command line. If this option is given then SoX will keep reading audio until it reaches the end of the input file. Intended for use when combining multiple input files, this option adjusts the volume of the file that follows it on the command line by a factor of FACTOR.

This is a linear amplitude adjustment, so a number less than 1 decreases the volume and a number greater than 1 increases it. If a negative number is given then in addition to the volume adjustment, the audio signal will be inverted.

See also the norm , vol , and gain effects, and see Input File Balancing above. The number of bits a. Not necessary with encodings that have a fixed number of bits, e. For example. By default i. The number of audio channels in the audio file. This can be any number greater than zero.

Note that if the file does in fact have two channels, this will result in the file playing at half speed. For an output file, this option provides a shorthand for specifying that the channels effect should be invoked in order to change if necessary the number of channels in the audio signal to the number given.

For example, the following two commands are equivalent:. The audio encoding type. Sometimes needed with file-types that support more than one encoding type. The available encoding types are as follows: signed-integer. A value of 0 represents minimum signal power. PCM data stored as unsigned integers.

Commonly used with an 8-bit encoding size. A value of 0 represents maximum signal power. International telephony standard for logarithmic encoding to 8 bits per sample. North American telephony standard for logarithmic encoding to 8 bits per sample. OKI a. IMA a. It utilises several audio formats with different bit-rates and associated speech quality. Encoding names can be abbreviated where this would not be ambiguous; e.

For example, if audio was recorded with a sample-rate of say 48k from a source that played back a little, say 1. For an output file, this option provides a shorthand for specifying that the rate effect should be invoked in order to change if necessary the sample rate of the audio signal to the given value.

Gives the type of the audio file. It can also be used to override the type implied by an input filename extension, but if overriding with a type that has a header, SoX will exit with an appropriate error message if such a header is not actually present. See soxformat 7 for a list of supported file types.

Endianness applies only to data encoded as floating-point, or as signed or unsigned integers of 16 or more bits. It is often necessary to specify one of these options for headerless files, and sometimes necessary for otherwise self-describing files. A given endian-setting option may be ignored for an input file whose header contains a specific endianness identifier, or for an output file that is actually an audio device. Specifies that the nibble ordering i.

See also N. Specifies that the bit ordering of the samples should be reversed; sometimes useful with a few mostly headerless formats. Output File Format Options These options apply only to the output file and may precede only the output filename on the command line. Specify the comment text to store in the output file header where applicable.

Specify a file containing the comment text to store in the output file header where applicable. The compression factor for variably compressing output file formats. If this option is not given then a default compression factor will apply.

The compression factor is interpreted differently for different compressing file formats. See the description of the file formats that use this option in soxformat 7 for more information. Note that applying multiple effects in real-time i. Stopping other applications may alleviate performance issues should they occur.

Multiple Effects Chains A single effects chain is made up of one or more effects. Audio from the input runs through the chain until either the end of the input file is reached or an effect in the chain requests to terminate the chain. SoX supports running multiple effects chains over the input audio.

In this case, when one chain indicates it is done processing audio, the audio data is then sent through the next effects chain. This continues until either no more effects chains exist or the input has reached the end of the file. An effects chain is terminated by placing a : colon after an effect. Any following effects are a part of a new effects chain. It is important to place the effect that will stop the chain as the first effect in the chain.

This is because any samples that are buffered by effects to the left of the terminating effect will be discarded. Further information on stopping effects can be found in the Stopping SoX section. There are a few pseudo-effects that aid using multiple effects chains.

These include newfile which will start writing to a new output file before moving to the next effects chain and restart which will move back to the first effects chain. Pseudo-effects must be specified as the first effect in a chain and as the only effect in a chain they must have a : before and after they are specified. The following is an example of multiple effects chains. It will split the input file into multiple files of 30 seconds in length. Each output filename will have unique number in its name as documented in the Output Files section.

Where applicable, default values for optional parameters are shown in parenthesis. The following parameters are used with, and have the same meaning for, several effects: center [ k ]. Used to specify the band-width of a filter. A number of different methods to specify the width are available though not all for every effect.

One of the characters shown may be appended to select the desired method as follows:. For each effect that uses this parameter, the default method i. Most effects that expect an audio position or duration in a parameter, i. The t suffix is entirely optional however, see the silence effect for an exception.

Note that the component values do not have to be normalized; e. Apply a two-pole all-pass filter with central frequency in Hz frequency , and filter-width width. The filter is described in detail in [1]. Apply a band-pass filter. The frequency response drops logarithmically around the center frequency. The width parameter gives the slope of the drop.

See also sinc for a bandpass filter with steeper shoulders. Apply a two-pole Butterworth band-pass or band-reject filter with central frequency frequency , and 3dB-point band-width width. The filters roll off at 6dB per octave 20dB per decade and are described in detail in [1]. Apply a band-reject filter.

See the description of the bandpass effect for details. This is also known as shelving equalisation EQ. Beware of Clipping when using a positive gain. If desired, the filter can be fine-tuned using the following optional parameters:.

The default value is Hz for bass or 3 kHz for treble. See also equalizer for a peaking equalisation effect. Changes pitch by specified amounts at specified times.

Each given triple: start-position , cents , end-position specifies one bend. The other values specify the points in time at which to start and end bending the pitch, respectively.

The pitch-bending algorithm utilises the Discrete Fourier Transform DFT at a particular frame rate and over-sampling rate. For example, an initial tone is generated, then bent three times, yielding four different notes in total:. Here, the first bend runs from 0.

Apply a biquad IIR filter with the given coefficients. Invoke a simple algorithm to change the number of channels in the audio signal to the given number CHANNELS : mixing if decreasing the number of channels or duplicating if increasing the number of channels. Add a chorus effect to the audio. This can make a single vocal sound like a chorus, but can also be applied to instrumentation.

Chorus resembles an echo effect with a short delay, but whereas with echo the delay is constant, with chorus, it is varied 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. See [3] for more discussion of the chorus effect.

Gain-out is the volume of the output. A typical delay is around 40ms to 60ms; the modulation speed is best near 0. For example, a single delay:. Compand compress or expand the dynamic range of the audio.

The attack and decay parameters in seconds determine the time over which the instantaneous level of the input signal is averaged to determine its volume; attacks refer to increases in volume and decays refer to decreases. For most situations, the attack time response to the music getting louder should be shorter than the decay time because the human ear is more sensitive to sudden loud music than sudden soft music.

Typical values are 0. The input values must be in a strictly increasing order but the transfer function does not have to be monotonically rising. If omitted, the value of out-dB1 defaults to the same value as in-dB1 ; levels below in-dB1 are not companded but may have gain applied to them.

The point 0,0 is assumed but may be overridden by 0, out-dBn. If the list is preceded by a soft-knee-dB value, then the points at where adjacent line segments on the transfer function meet will be rounded by the amount given. The third optional parameter is an additional gain in dB to be applied at all points on the transfer function and allows easy adjustment of the overall gain.

The fourth optional parameter is an initial level to be assumed for each channel when companding 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 action has begun to operate: it is quite probable that in such an event, the output would be severely clipped while the compander gain properly adjusts itself. The fifth optional parameter is a delay in seconds. The input signal is analysed immediately to control the compander, but it is delayed before being fed to the volume adjuster.

A typical value is 0. The following example might be used to make a piece of music with both quiet and loud passages suitable for listening to in a noisy environment such as a moving vehicle:. In the next example, compand is being used as a noise-gate for when the noise is at a lower level than the signal:. Here is another noise-gate, this time for when the noise is at a higher level than the signal making it, in some ways, similar to squelch :.

See also mcompand for a multiple-band companding effect. Comparable with compression, this effect modifies an audio signal to make it sound louder. See also the compand and mcompand effects. Apply a DC shift to the audio. 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 stat or stats effect can be used to determine if a signal has a DC offset. An optional limitergain can be specified as well. It should have a value much less than 1 e. An alternative approach to removing a DC offset albeit with a short delay is to use the highpass filter effect at a frequency of say 10Hz, as illustrated in the following example:. Pre-emphasis was applied in the mastering of some CDs issued in the early s.

These included many classical music albums, as well as now sought-after issues of albums by The Beatles, Pink Floyd and others. Pre-emphasis should be removed at playback time by a de-emphasis filter in the playback device. However, not all modern CD players have this filter, and very few PC CD drives have it; playing pre-emphasised audio without the correct de-emphasis filter results in audio that sounds harsh and is far from what its creators intended.

With the deemph effect, it is possible to apply the necessary de-emphasis to audio that has been extracted from a pre-emphasised CD, and then either burn the de-emphasised audio to a new CD which will then play correctly on any CD player , or simply play the correctly de-emphasised audio files on the PC.

The de-emphasis filter is implemented as a biquad and requires the input audio sample rate to be either Maximum deviation from the ideal response is only 0. See also the bass and treble shelving equalisation effects. Delay one or more audio channels such that they start at the given position. For example, delay 1. The following one long command plays a chime sound:.

Apply dithering to the audio. Dithering deliberately adds a small amount of noise to the signal in order to mask audible quantization effects that can occur if the output sample size is less than 24 bits. With no options, this effect will add triangular TPDF white noise. Note that most filter types are available only with Hz sample rate. The filter types are distinguished by the following properties: audibility of noise, level of inaudible, but in some circumstances, otherwise problematic shaped high frequency noise, and processing speed.

The most likely use for this is when applying fade in or out to an already dithered file, so that the redithering applies only to the faded portions. However, auto dithering is not fool-proof, so the fades should be carefully checked for any noise modulation; if this occurs, then either re-dither the whole file, or use trim , fade , and concatencate. This effect should not be followed by any other effect that affects the audio.

Downsample the signal by an integer factor: Only the first out of each factor samples is retained, the others are discarded. No decimation filter is applied. If the input is not a properly bandlimited baseband signal, aliasing will occur.

This may be desirable, e. For a general resampling effect with anti-aliasing, see rate. See also upsample. Makes audio easier to listen to on headphones.

Add echoing to the audio. 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. Multiple echoes can have different delays and decays. Each given delay decay pair gives the delay in milliseconds and the decay relative to gain-in of that echo.

For example: This will make it sound as if there are twice as many instruments as are actually playing:. If the delay is very short, then it sound like a metallic robot playing music:. Add a sequence of echoes to the audio. Each delay decay pair gives the delay in milliseconds and the decay relative to gain-in of that echo. Care should be taken using many echos; a single echos has the same effect as a single echo.

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 band-pass and band-reject filters that at all other frequencies is unchanged.

In order to produce complex equalisation curves, this effect can be given several times, each with a different central frequency. See also bass and treble for shelving equalisation effects. The default is logarithmic. A fade-in starts from the first sample and ramps the signal level from 0 to full volume over the time given as fade-in-length. Specify 0 if no fade-in is wanted. For fade-outs, the audio will be truncated at stop-position and the signal level will be ramped from full volume down to 0 over an interval of fade-out-length before the stop-position.

If fade-out-length is not specified, it defaults to the same value as fade-in-length. No fade-out is performed if stop-position is not specified. Any time specification may be used for fade-in-length and fade-out-length. Apply a flanging effect to the audio. See [3] for a detailed description of flanging. Apply amplification or attenuation to the audio signal, or, in some cases, to some of its channels. Without other options, gain-dB is used to adjust the signal power level by the given number of dB: positive amplifies beware of Clipping , negative attenuates.

With other options, the gain-dB amplification or attenuation is logically applied after the processing due to those options. For example,. Note that limiting more than a few dBs more than occasionally in a piece of audio is not recommended as it can cause audible distortion.

See the compand effect for a more capable limiter. For example, with. Of course, with bass, it is obvious how much headroom will be needed, but with other effects e. The above effects chain guarantees never to clip nor amplify; it attenuates if necessary to prevent clipping, but by only as much as is needed to do so. Here, the second gain invocation, reclaims as much of the headroom as it can from the preceding effects, but retains as much headroom as is needed for subsequent processing.

Apply a high-pass or low-pass filter with 3dB point frequency. The filters roll off at 6dB per pole per octave 20dB per pole per decade.

The double-pole filters are described in detail in [1]. See also sinc for filters with a steeper roll-off. Apply an odd-tap Hilbert transform filter, phase-shifting the signal by 90 degrees. This is used in many matrix coding schemes and for analytic signal generation. The process is often written as a multiplication by i or j , the imaginary unit. An odd-tap Hilbert transform filter has a bandpass characteristic, attenuating the lowest and highest frequencies.

By default, the number of taps is chosen for a cutoff frequency of about 75 Hz. The first argument is the plugin module, the second the name of the plugin a module can contain more than one plugin , and any other arguments are for the control ports of the plugin. Missing arguments are supplied by default values if possible. Normally, the number of input ports of the plugin must match the number of input channels, and the number of output ports determines the output channel count. Some plugins introduce latency which SoX may optionally compensate for.

Loudness control – similar to the gain effect, but provides equalisation for the human auditory system. The gain is adjusted by the given gain parameter usually negative and the signal equalised according to ISO w. Apply a low-pass filter. See the description of the highpass effect for details. The multi-band compander is similar to the single-band compander but the audio is first divided into bands using Linkwitz-Riley cross-over filters and a separately specifiable compander run on each band.

See the compand effect for the definition of its parameters. Compand parameters are specified between double quotes and the crossover frequency for that band is given by crossover-freq ; these can be repeated to create multiple bands. For example, the following one long command shows how multi-band companding is typically used in FM radio:.

The audio file is played with a simulated FM radio sound or broadcast signal condition if the lowpass filter at the end is skipped. Note that the pipeline is set up with US-style 75us pre-emphasis. See also compand for a single-band companding effect. Calculate a profile of the audio for use in noise reduction. See the description of the noisered effect for details. Reduce noise in the audio signal by profiling and filtering.

This effect is moderately effective at removing consistent background noise such as hiss or hum. To use it, first run SoX with the noiseprof effect on a section of audio that ideally would contain silence but in fact contains noise – such sections are typically found at the beginning or the end of a recording. How much noise should be removed is specified by amount – a number between 0 and 1 with a default of 0.

Higher numbers will remove more noise but present a greater likelihood of removing wanted components of the audio signal. Before replacing an original recording with a noise-reduced version, experiment with different amount values to find the optimal one for your audio; use headphones to check that you are happy with the results, paying particular attention to quieter sections of the audio. On most systems, the two stages – profiling and reduction – can be combined using a pipe, e.

Normalise the audio. Out Of Phase Stereo effect. Mixes stereo to twin-mono where each mono channel contains the difference between the left and right stereo channels.

So your saying, let the Lumin do the upscaling? I use the upsample all my files to DSD before with some Izotope Settings then, with sliders settings that people says were good… but it is all different from house to house, system to system… so you have to adjust that yourself and compare to really see a difference?

Please clarify or am I missing something? Sound quality wise, no. Jdelena many people in this forum prefer not to upsample and fine the sound more natural. In my situation, I spent a lot of time with or without upsampling, going back and forth again and again. I ended up with Audirvana upsampling before sending the data to the DAC. It is really a matter of personal choice. With regard to bugs, I have been lucky and I never had any problems with my settings.

 
 

Sox settings audirvana free

 
replace.me › Forums › Equipment › Software. I’ll try to find a free FLAC which benefits greatly from upsampling, and upload upscaled variations with different settings for others to try. Several PS Audio forum members seem to be familiar with or are using Audirvana Plus 3. might suggest settings for the iZoptope and/or SOX audio filters. From.

 

Sox settings audirvana free –

 
You can lose a lot of time in it. Did you read Archimago’s article that Mitchco referenced?