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Sound cards section
Configuration of input and output sound cards, and synchronization.

 Besides choosing sound cards, this section also lets you configure FM transmitter synchronization, which can be used to synchronize the sound at multiple FM transmitter sites, using a normal Shoutcast or other stream as input.

 A few notes:
  • If you want low latency audio, use ASIO for both input and output. If you don't use ASIO, Windows gives you an extra 100-300 ms of latency, which is far too much if you are listening to yourself for example on a headphone.
  • If you use multiple sound cards which don't share the same clock, the output buffer will at some point get underruns (causing drops in audio) or overruns (causing some pieces of audio to get lost). See Synchronize to output for a (partial) solution.


 I/O panel

 Sound cards panel

 ASIO panel

 Display synchronization panel
Synchronize the metering to the audio.
  • Synchronize display with
    Select what to synchronize the display to.

     If you choose input, events are shown as fast as possible, to match the incoming levels even if the output is sent out with a delay.

 Restart sound cards panel
Restart the whole sound card section.
  • Restart sound cards
    Restarts the whole sound card section. Try this is you are running into

     trouble (buffer underruns, sound cards that refuse to connect).

     Normally, in case of trouble this is done automatically, so you should never need this.
Sample rate section
Selects the sample rate used for input and output.

 General panel
Sample rate settings.
  • Sample rate
    The sample rate to use for input.

     Normally, the output sample rate will be identical to the input sample rate. The only exception is when you are not using ASIO, the samle rate is lower than 128 kHz and FM output is used.

     If you use samle rates above 48 kHz, the audio will be downsampled before processing and upsamled again afterwards. 88.2 and 176.4 kHz are downsampled to 44.1 kHz, 96 and 192 kHz are downsampled to 48 kHz.

     The sample rate has a small effect on the latency and CPU load. At 48 kHz there is a bit more data that needs to be processed, which increases the CPU load slightly. The latency is a bit smaller because a block of the same number of samples is a bit smaller. If you use 32 kHz or a multiple thereof, the CPU load is a lot smaller - but you should only use it if you don't need any audio above approx. 14 kHz.

     If you want to use FM output with stereo and RDS encoding, and you are using ASIO, then the sample rate must be set to at least 128 kHz - 176.4 or 192 are preferred. If you don't use ASIO for the output, the output sound card will automatically be opened with a high enough sample rate to send out the stereo and RDS signals.

     Important: If you are not using ASIO and you are using Windows Vista, 7 or 8, you need to make sure that the sound card driver is configured to use the same sample rate that you are setting in Stereo Tool, otherwise Windows will resample it, which causes artifacts any may cause the FM output to not work at all. Go to Controls Panel -> Sound -> The sound card that you want to use -> Advanced -> Standard setting.
ASIO section
Use ASIO for input and output. Use this if possible!

 This gives better control over the sound card leading to less potential problems, and it greatly reduces the delay between input and output.

 ASIO panel
The ASIO settings.
  • ASIO Device ID
    The ASIO device to be used.

     Only one ASIO device can be used in a program at once.

    •
  • ASIO block size
    Override the default ASIO block size.

     The ASIO block size is typically set by the ASIO driver. In most cases it's possible to overrule the ASIO block size, to get lower latencies, or to use bigger blocks to improve stability. Too small block sizes can cause very brief clicks in the audio, depending on the CPU, Windows version and sound card driver).

     Note that some drivers don't allow using different block sizes, and not all sizes work on all devices. If a selected block size is not accepted by the driver, Stereo Tool will try to increase it until it is accepted.

    •
  • Mask ASIO buffer underruns (clicking)
    Protects against clicks and pops due to buffer underruns.

     When the ASIO buffer size is set very small, to minimize latency, in some cases a block of audio may not yet be available when it needs to be sent to the ASIO driver, which normally results in audible clicks.

     This setting lets Stereo Tool predict the data that would be sent to the sound card, and lets it send that instead. This masks most of the clicks and pops, but not entirely.

     It would make sense to have this option enabled at all times, but to more easily hear when the buffer setting is correct, it is advised to turn this checkbox off when you are setting the buffer sizes, and on again afterwards as some extra protection.

    •
  • Open ASIO Control Panel
    Fire up the sound card driver's ASIO control panel.

     The ASIO control panel is a window provided by the sound card driver which lets you configure certain things, such as the ASIO block size (which affects latency).

     If you want an as low as possible latency, find the lowest latency value here that works without hiccups.

     Not all sound card drivers support this button (although you can usually change the settings elsewhere if they don't).

    •
  • Try to fix Marian driver bug
    The Marian Trace Alpha sound card driver for Windows 7

     (probably also Windows Vista and 8) causes some systems to lock up completely after running for some time - typically a few days.

     The cause of this appears to be something that's timing-related: If the ASIO code in Stereo Tool returns control to the driver very quickly, it happens more frequently (within a few hours instead of days).

     If this slider is set higher, the time before control is given back to the sound card driver is increased, which seems to completely remove the lock-ups.

     For non-Marian cards, and for Marian cards on Windows XP, you can safely set this slider to 0 (which slightly reduces the CPU load). The default setting (3) fixes the problem for all the people who have reported this issue so far; if you run into it anyway, you can try a higher value. Please contact us if this does not fix the issue.

    •
  • Always open all ASIO channels

 ASIO monitoring panel
  • Recovered glitches

    •
  • Not recovered glitches

    •
  • Reset glitch counters

 Input panel
Main input sound card settings.
  • Left
    ASIO left channel input.

     If ASIO is enabled, selecting an ASIO input port here overrules the setting in Main input. As soon as an ASIO input port is selected, ASIO will be used and the buffer filling display will turn from blue to green.

    •
  • Right
    ASIO right channel input.

     See Left. You can select two sound card input channels for stereo input.

 Input #2 panel
Second audio input, mostly used for SCA audio encoded at 67 kHz.

 HD Output panel
Sound card settings for normal (non-FM) output.

 FM Output panel
Sound card settings for FM output.

 Note that if you use ASIO for FM output and the sample rate is not at least 176.4 kHz, some parts of the FM spectrum cannot be sent to the sound card.

 Low Latency Monitoring Output panel
Sound card settings for Low Latency Monitoring Output.
  • Override MME safety buffer

    •
  • Override default MME safety buffer

    •
  • Override DirectSound safety buffer

    •
  • Override default DirectSound safety buffer

    •
  • Override WASAPI safety buffer

    •
  • Override default WASAPI safety buffer

    •
  • Override Kernel Streaming safety buffer

    •
  • Override default Kernel Streaming safety buffer
Input section
Input audio settings.

 You can receive input using ASIO (preferred if your sound card supports it), the standard Windows audio layer, and if you have installed VLC, an audio stream.

 Input panel
  • Main input
    Sound card to use for audio input.

     If this is set to Streaming (via VLC), a stream is used for input instead. This setting can be overruled by Left and Right.

 Sound card driver issues workaround, use only if you get constant glitches panel

 Input sound card panel

 More input sound card settings panel

 Levels panel

 Tilt correction panel
  • Overrule I/O block size

    •
  • Block size (@44/48)

    •
  • Stream URL
    Stream address, if Main input is set to Stream (via VLC)

     A valid address of a stream that VLC can decode. This requires that VLC (32 bit version for 32 bit Stereo Tool, 64 bit version for 64 bit Stereo Tool) is installed on your system. Currently only works in Windows.

     If you have a problem with a stream, please try if you can open it in VLC Media Player directly.

     If necessary you can supply extra VLC command line arguments after the URL. For example --extraintf logger -vvv will show a logging window that might be useful to find out what's happening exactly (do not turn this on during normal usage because it can make things unstable). Other options can be used to control sample rate, buffering and more. For a full list of options (some can cause big problems when using them from within Stereo Tool) see VLC command line help.

 AES67 panel
AES67 input.

 With this, you can read AES67 (or other RTP) streams.
  • Subrange

    •
  • NIC
    Selection of the input network interface.

 AES67 discovery panel
  • Enable Livewire discovery

    •
  • Enable Dante discovery

    •
  • Enable Ravenna discovery

    •
  • Device

    •
  • Channel

    •
  • Port (239.69.*:5004)

    •
  • Maximum jitter
    Maximum time after the expected time of receiving an AES67 packet to give up and insert silence.

     This value must be higher than all the output buffer sizes to avoid problems when there are network issues.

 Low input level correction panel
Adjusts the input level if the input level is low.
  • Gain
    Adjusts the input level to reach around 0 dB peak level.

     Many studios use a lot of headroom in their signal, they feed the processor at levels like -24 dB. The built-in presets in Stereo Tool were designed for input that reaches levels upto about 0 dB regularly. If you feed the audio at -24 dB, certain filters (Declipper, Noise gate, AGC) don't function properly and the sound will be bad.

     Of course, it should still be possible for studios to use a large amount of headroom. With this slider you can adjust the level such that under normal circumstances the peaks are at about 0 dB. If peaks are occasionally louder, that's no problem - no cliping is performed and no distortion is created.

    •
  • Balance
    Adjusts for different input levels on left and right channel.

     If the left and right channel input levels aren't perfectly equal (which might happen for example if you're using some analog equiment), this slider allows you to correct it. A negative value means that the left channel is boosted, a positive value means that the right channel is boosted. Don't look too much at the value, instead play some mono audio and adjust this slider until the input levels (which you can see in the VU meters) are equal for left and right.

     Most other devices (including amplifiers) will reduce the level of a channel instead of increasing it. The reason to increase the level is that otherwise it's more difficult to show both the original and the modified (by Gain and Balance) volume levels - one channel could be made louder while the other could be made softer.

    •
  • Channel selection
    Selects which of the input channels to use for left and right audio.

     This can be useful if you want to use only one input channel (mono), or if you want to swap left and right.

 Synchronize input to output sound cards panel
Avoids buffer underruns or overruns if different sound cards are used for input and output.
  • Synchronize to output
    Resamples the input to match the output sample rate.

     If you use a different sound card for input and output, chances are that the sample rates do not match perfectly and after some time (hours to days) the output buffers suffer from underruns (buffer is empty, moment of silence gets inserted) or overruns (buffer is too full, input is ignored and some audio is lost).

     The same problem occurs if you feed the input through a virtual audio cable (VAC, VB Cable).

     Enabling this setting matches the input sample rate to the output sample rate.

     Note that this uses extra processing power, so if it is not needed you should keep it turned off.

 Advanced synchronisation settings panel
  • Relative adjust
    Controls how fast synchronization works

     Synchronization is done by slightly changing the sample rate of the input signal. This means that it has a (small) effect on the pitch of audio. Using a faster speed means that synchronization works faster, but gets more audible.

     The default setting is 1%, which means that the maximum possible adjustment if things are really wrong is 1%.

    •
  • Resampling quality
    Quality of the resampling algorithm. Affects CPU load and quality.

 Input tilt panel
Corrects tilt problems in the input signal.

 If you have an analog signal path from the studio to Stereo Tool, you might have some tilt issues (a square wave does not look like a square wave anymore.

Tilt correction lets you correct this, which slightly improves the audio quality (mainly the bass) and also helps the declipper to function optimally.
  • Tilt correction enabled
    Enables Tilt correction.

    •
  • Lows tilt correction (RC)
    RC value of the first highpass filter.

     Tilt is usually caused by a Resistor and Capacitor (RC) circuit, which acts as a highpass filter. For example, a sound card might use an RC circuit to remove DC offset from a signal. Unfortunately, this also reduces the audio quality and makes processing more difficult.

     The tilt correction filter in Stereo Tool inverts the effect of an RC circuit, thus restoring the original signal (except for the DC offset).

     This slider sets the RC value of the RC circuit to be inverted.

    •
  • Second tilt correction different from first
    Makes Override second lows tilt correction stage setting identical to Lows tilt correction (RC).

     In all the cases that we have seen so far, this gives the best results.

    •
  • Override second lows tilt correction stage
    Second RC circuit inversion value

     See Lows tilt correction (RC). This slider allows to invert a second RC circuit.

     Note: In many cases, the best results are obtained by using the same value for both RC inversion sliders.

    •
  • Fade speed
    Protection against restoring DC offsets.

     To avoid restoring a DC offset, if the inversion of the RC circuits causes a large offset, this slider controls how fast this DC offset is removed. Default value is 100%, you should probably leave it there.

 Input & detilted input panel
Display of the input before (left) and after (right) detilting.
Input #2 section
Second audio input, mostly used for SCA audio encoded at 67 kHz.

 Note that this is NOT intended to feed an external RDS encoder.

 Input #2 panel
SCA sound card settings.
  • Input 2
    Enable the secondary (usually SCA) input.

    •
  • Input 2
    Sound card to use for SCA audio input.

     If this is set to Streaming (via VLC), a stream is used for input instead. This setting can be overruled by Link error '519' and Link error '520'.

     If the input is coming from a different sound card than Main input, hiccups may occur on one of the inputs.

 Input #2 usage panel
Use the Input #2 for other purposes than SCA.
  • Use SCA Input as external RDS input
    Enables external RDS input via Input #2.

     Internally, the pilot will be synchronized to the incoming RDS signal - any incoming pilot signal will be ignored. If this RDS signal is of really bad quality, it might cause problems, and if the RDS signal is weak the internal stereo pilot generator will take over. Except for that, the input level does not matter.

    •
  • No SCA, add this sound card input to normal Input
    Combine both inputs and use the result as processing input, or setup a backup input.

     This can be used either for providing a backup input and switching to it in case of signal loss, or to combine signals from multiple studio's without electrically coupling them.

    •
  • Use as Backup Input
    Don't combine the inputs, but use Input #2 as backup.

    •
  • Use for

    •
  • Threshold level
    Main input silence detection threshold level.

     Only used if Don't combine the inputs, but use Input #2 as backup. is checked.

    •
  • Minimum silence
    Minimum time of detecting silence before switching to Input #2.

     Only used if Don't combine the inputs, but use Input #2 as backup. is checked.

 Low input level correction panel
Adjusts the input level if the input level is low.
  • Gain
    Adjusts the input level to reach around 0 dB peak level.

     Many studios use a lot of headroom in their signal, they feed the processor at levels like -24 dB. The built-in presets in Stereo Tool were designed for input that reaches levels upto about 0 dB regularly. If you feed the audio at -24 dB, certain filters (Declipper, Noise gate, AGC) don't function properly and the sound will be bad.

     Of course, it should still be possible for studios to use a large amount of headroom. With this slider you can adjust the level such that under normal circumstances the peaks are at about 0 dB. If peaks are occasionally louder, that's no problem - no cliping is performed and no distortion is created.

 Synchronize input 2 to output sound cards panel
  • Synchronize to output

 Advanced synchronisation settings panel
  • Relative adjust

    •
  • Resampling quality

    •
  • SPEED (breaks audio, can cause asserts)

 Input 2 tilt panel
Corrects tilt problems in the input signal.

 If you have an analog signal path from the studio to Stereo Tool, you might have some tilt issues (a square wave does not look like a square wave anymore.

Tilt correction lets you correct this, which slightly improves the audio quality (mainly the bass) and also helps the declipper to function optimally.
  • Correction enabled
    Enables Tilt correction.

    •
  • RC
    RC value of the first highpass filter.

     Tilt is usually caused by a Resistor and Capacitor (RC) circuit, which acts as a highpass filter. For example, a sound card might use an RC circuit to remove DC offset from a signal. Unfortunately, this also reduces the audio quality and makes processing more difficult.

     The tilt correction filter in Stereo Tool inverts the effect of an RC circuit, thus restoring the original signal (except for the DC offset).

     This slider sets the RC value of the RC circuit to be inverted.
HD Output section
Settings for normal (non-FM) output.

 If FM processing is used, then the normal output contains the signal after demodulating and de-emphasis. It should be similar to the sound that people hear on their FM radios.

 Normal panel
Sound card settings for normal (non-FM) output.
  • HD output
    Enables the normal (non-FM) output.

    •
  • HD output
    Sound card to which the normal (non-FM) output should be sent.

     If this is set to Streaming (via VLC), a stream is used for input instead. This can be overruled by Link error '511' and Link error '512'.

 Sound card driver issues workaround, use only if you get constant glitches panel

 More HD Output settings panel
  • Overrule I/O block size

    •
  • Block size (@44/48)

    •
  • Separate processing paths active for FM and non-FM

    •
  • Signal selection
    Selection of the sound that will be played though Normal.

     If FM Output plays FM or AM transmitter output, with this setting you can select what Normal will play.

     If you select a separately processed output, clipping and limiting will be done separately for the FM Output and Normal outputs. This means that you can generate a signal suitable for streaming without suffering from the loss in high frequency content caused by FM pre-emphasis, for example.

     Options are:
    Input without processing  Plays the original input. Can be useful for monitoring.
    De-emphasized version of FM outputWhat listeners will hear on an FM receiver.
    Separately processed streaming inputSeparate clipping and limiting for the stream.
    Separately processed pre-emphasized left/right FM outputPre-emphasized stereo FM output, without composite clipper.
    Separately processed de-emphasized left/right FM outputDe-emphasized stereo FM output, without composite clipper.


     If separate processing is needed, the CPU load will go up. When that's the case, the botton next to the Signal selection pulldown menu will be lit.

    •
  • Volume
    Normal (non-FM) output volume.

    •
  • Buffer size / Diversity Delay
    Normal (non-FM) output buffer size.

     Using a bigger buffer size reduces the chance of getting audio dropouts if the pc is very busy, but it also increases the delay between input and output. If you want to do other things on the pc that's running Stereo Tool (starting other programs while processing is running etc.), you might need a big buffer here.

     If you use ASIO (Link error '511', Link error '512'), the buffer size can be much lower than if you're not. If you're not using ASIO there are also buffers in Windows that add a lot of extra delay.

     To get the lowest possible latency (see also Processing latency), set this value as low as possible without getting audio drops, and avoid running other programs that cause hiccups.

 Restart on buffer issues panel
Controls buffer restarting.

 If the sound card buffer is nearly empty or nearly full for a longer period of time, which indicates that some sort of problem has occurred and might lead to hiccups in the audio, this will automatically act as if Restart sound cards is pressed.
  • Auto-restart on buffer issues
    Enables the automatic restart feature.

    •
  • Restart if buffer filling gets below
    Restarts if the output buffer is filled below this percentage for a longer period.

    •
  • Restart if buffer filling gets above
    Restarts if the output buffer is filled above this percentage for a longer period.

 Synchronize HD output to FM output panel
Synchronizes the Normal to the FM Output.

 If different sound cards are used for FM and non-FM (streaming, HD, DAB, listening) outputs, they will drift after a while, which leads to hiccups or lost audio. By enabling synchronization, the Normal will be synchronized to match the FM Output sample rate. This is also needed when one of the sound cards is a virtual cable.
  • Synchronize to FM or input
    Enables synchronization.

     If this is set to Auto, Stereo Tool will compare sound card names to 'guess' if synchronization is needed. If the latency is very low, Auto will never enable synchronization.

 Advanced synchronisation settings panel
  • Extra delay (added to buffer size)
    Extra delay for synchronization.

     This can be used for example to synchronize a DAB or HD signal to the FM signal, with the amount of shift in time that's needed to make them arrive simultaneously at a receiver.
FM Output section
Settings for FM output.

 FM panel
Sound card settings for FM output.
  • FM output
    Enables FM output.

    •
  • FM output
    Sound card to which the normal (non-FM) output should be sent.

     This can be overruled by Link error '' and Link error ''.

 More FM output settings panel

 FM output calibration panel

 FM sound card output levels panel

 Sound card driver issues workaround, use only if you get constant glitches panel
  • Overrule I/O block size

    •
  • Block size (@44/48)

    •
  • VLC SOUT=
    VLC --sout string for stream encoding, if FM output is set to VLC.

     See Link error '2793'.

    •
  • Composite output level
    FM output volume. Need to be calibrated for a compliant FM signal!

    •
  • Separate channel 2
    Enables separate volume control for the 2nd (right) channel.

     Used when you have multiple transmitters connected to a single Stereo Tool instance.

    •
  • Separate second channel output level
    The relative volume of the right channel.

    •
  • Buffer size
    FM output buffer size.

     Using a bigger buffer size reduces the chance of getting audio dropouts if the pc is very busy, but it also increases the delay between input and output. If you want to do other things on the pc that's running Stereo Tool (starting other programs while processing is running etc.), you might need a big buffer here.

     If you use ASIO (Link error '', Link error ''), the buffer size can be much lower than if you're not. If you're not using ASIO there are also buffers in Windows that add a lot of extra delay.

     To get the lowest possible latency (see also Processing latency), set this value as low as possible without getting audio drops, and avoid running other programs that cause hiccups.

 Test signals panel
Shows the output before (left) and after (right) tilt correction.

 Important: You need to make sure that there's enough headroom for the tilt correction. So, if you feed a 15 Hz square wave, the waveform should not clip. If it does, you need to lower the output level to the sound card (and adjust the transmitter to handle the lower level).
  • Generate test tone
    Send out test tones to calibrate an FM transmitter.

    •
  • Test tone

    •
  • Frequency
    Test tone frequency.

     The frequency of the test tone if the Link error 'type' is set to Sine, Square or Smooth Square. Otherwise this slider is ignored.

 FM Tilt correction panel
Corrects tilt caused by sound card, FM transmitter or cables.

 Almost all sound cards (except those by Marian) use a highpass filter in their output to remove and DC offset from the signal. This causes overshoots when broadcasting a tightly clipped MPX signal. Similarly, some transmitters slowly adjust their frequency when there's a DC offset, which has a similar effect.

 This filter reverses the effect to make it look like there is no RC circuit at all.

 Some values measured with this filter on the built-in sound card of a laptop:
Tranmitter with the volume calibrated to give 75 kHz modulation for a 1000 Hz sine wave at maximum level:
  • 15 Hz sine wave: 74 kHz modulation
  • 1000 Hz sine wave: 75 kHz modulation
  • 60000 Hz sine wave: 73 kHz modulation
  • 15 Hz square wave: 91 kHz modulation
After calibrating Tilt Correction:
  • 15 Hz sine wave: 75 kHz modulation
  • 1000 Hz sine wave: 75 kHz modulation
  • 60000 Hz sine wave: 73 kHz modulation
  • 15 Hz square wave: 75 kHz modulation
This YouTube video describes how to set it up:

 See this YouTube video for a detailed explanation of how to set this up correctly:


  • Correction enabled
    Enables input tilt correction.

    •
  • Lows tilt correction (RC)
    Tilt correction value.

     Tilt is usually caused by a Resistor and Capacitor (RC) circuit, which acts as a highpass filter. For example, a sound card might use an RC circuit to remove DC offset from a signal. Unfortunately, this also reduces the audio quality and makes processing more difficult.

     The tilt correction filter in Stereo Tool inverts the effect of an RC circuit, thus restoring the original signal (except for the DC offset).

     This slider sets the RC value of the RC circuit to be inverted.

    •
  • Second tilt correction different from first
    Makes Override second lows tilt correction stage value identical to Lows tilt correction (RC).

     In all the cases that we have seen, using identical values gives the best result.

    •
  • Override second lows tilt correction stage
    Second RC circuit inversion value

     See Lows tilt correction (RC). This slider allows to invert a second RC circuit.

    •
  • Fade speed
    Protection against restoring DC offsets.

     To avoid restoring a DC offset, if the inversion of the RC circuits causes a large offset, this slider controls how fast this DC offset is removed. Default value is 100%, you should probably leave it there.

    •
  • High frequency dropoff
    Enables Highs correction.

     Some sound cards have a lowpass filter that affects the highest frequencies that are needed for FM (the stereo/RDS area). This checkbox enables compensation for that.

    •
  • Highs dropoff
    Highs correction value.

     See High frequency dropoff. To properly set up this value, a good quality analyser is needed - many analyzers aren't completely flat themselves and are not suitable, in those cases it's probably safer to just stick with what the sound card does.

 More tilt correction settings panel

 Synchronization between different sound cards panel

 Latency settings panel

 OS safety settings panel

 Output buffers panel

 Synchronize FM transmitters panel
Synchronizes audio on FM transmitters using standard Shoutcast etc. streams.

 If an FM or AM station has multiple transmitters and RDS is used to automatically switch between those frequencies, it's very important that the signals are synchronized so that switching between the frequencies is not noticeable for a listener.

 This is often achieved using specific hardware on both the sending (studio) and receiving (transmitter) end. This hardware is usually expensive, and it would be much simpler if a station could just use an exising (free) streaming protocol to send the audio to the transmitters - there's already a PC there that runs Stereo Tool which could receive the signals.

 The problem with most streaming mechanisms is that there is nothing built in to keep the signals synchronized. In fact, even if you start two players on the same pc, there's often a difference of multiple seconds between the two players. And due to minimal differences in sample rate between the sound cards in pc's, after a long period (weeks-months) the signal that comes out of the different pc's could be multiple minutes apart.

 Stereo Tool can synchronize any type of stream that you can feed to it, as long as it gets the audio as soon as it arrives (it should not be buffered elsewhere). This means that the plugin version of Stereo Tool can do it, and the stand alone version can do it if you have configured it to be able to receive streams.

This video shows how well it works on a station in Belgium
that uses Stereo Tool on their 13 FM frequencies.


  • Max dynamic adjustment
    Maximum amount of speed adjustment.

     See also Relative adjust. This limits the maximum amount of speed adjustment.

    •
  • Turbo (when opening device)
    Faster synchronization after startup of after loss of signal.

     During normal operation, the synchronization should be done very prudently. But directly after connecting or reconnecting to a stream, to avoid taking multiple hours to reach synchronization, it can be done a bit faster. Turbo controls how fast the speed can be adjusted in this situation. Note that the maximum amount of speedup is still controlled by Max dynamic adjustment, so if that is not set too high it should still be nearly unnoticeable. The Relative adjust effect is increased in Turbo mode, so there is no big slowdown when the target point is reached.

    •
  • Extra delay (added to buffer size)
    Compensation of small constant delay differences between sites.

     If there's a constant small difference in timing between sites (this is usually not the case), you can correct it here.

 Single Frequency Networks (SFN) panel
Sample-exact synchronization between 2 transmitters at the same frequency.
  • Delay one of the MPX outputs
    Enables SFN support.

    •
  • Channel
    Selects which channel to delay.

    •
  • Invert
    Phase inverts one of the channels.

    •
  • Delay
    Time to delay Channel.

     Time is measured in samples here.

 Restart on buffer issues panel
See Restart on buffer issues.

 Synchronization with HD panel
FM delay to synchronize with HD.
Extreme calibration settings section
This is obsolete and should normally not be needed anymore.

 Testing panel

 Frequency panel
Low Latency monitoring (lower quality) section
Settings for Low Latency Monitoring Output.

 Latency panel
Low Latency Monitoring output settings.
  • Low Latency output
    Enables the Low Latency Monitoring Output.

    •
  • Monitoring output
    The sound card to use for Low Latency Monitoring Output.

     Note that using Low Latency Monitoring Output without ASIO is useless because the delay that Windows adds is far too big.

    •
  • Volume
    The Low Latency Monitoring Output volume.

     Must be set low enough to handle spikes because no clipping is performed on this output.

    •
  • Processing latency
    Processing latency for the Low Latency monitoring (lower quality).

     Comparable to Processing latency, but for the low latency output.

     Quality 128 gives a processing latency of under 3 ms, allowing a total input-to-output latency of about 5-6 ms. At 256, the processing latency is a bit below 6 ms, with a total input-to-output latency of around 9 ms.

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  • Audio quality
    Enables higher quality Low Latency monitoring (lower quality).

     This setting greatly reduces artifacts in the low latency monitoring output. Turning this feature on doubles the CPU load of the low latency processing. It does not affect the other outputs.

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  • Buffer size
    Controls the delay. Set as low as possible.
I/O self test section
Built-in self test to check sound card behavior.

 The self test generates a report that can be used to determine if the sound card in the system behaves properly. Among others, it measures the sound card frequecy response, noise level and distortion.

 To use this,
  • In the Stereo Tool instance that will measure the output (this can be the same one that sends out the test tones, but it can also be a different instance on a different pc), first set up the Output folder where the report will be written to.
  • Then enable Check self test audio on that same pc.
  • Then, on the pc that will send the test tones, enable Send self test audio.


 Make sure that the receiving instance is listening to the correct output. The test will take several minutes, when it's finished a text file will be written in the output folder. The text file name contains the start date and time of the test.

 I/O Self test panel
  • Test type

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  • Glitch threshold