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.
Display synchronization panel Synchronize the metering to the audio.
Synchronize 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 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 Use ASIO. ASIO still needs to be enabled per input or output.
Only one ASIO device can be used in a program at once.
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).
Input panel Main input sound card settings.
ASIO Override input channel 1 ASIO left channel input.
If ASIO is enabled, selecting an ASIO input port here overrules the setting in Input Device ID. As soon as an ASIO input port is selected, ASIO will be used and the buffer filling display will turn from blue to green.
ASIO Override input channel 2 ASIO right channel input.
SCA panel Second audio input, mostly used for SCA audio encoded at 67 kHz.
ASIO Override input 2 channel 1 ASIO SCA input #1
If ASIO is enabled, selecting an ASIO input port here overrules the setting in Input 2 Device ID. As soon as an ASIO input port is selected, ASIO will be used and the buffer filling display will turn from blue to green.
Normal panel Sound card settings for normal (non-FM) output.
ASIO Override channel 1 ASIO left channel normal output.
If ASIO is enabled, selecting an ASIO output port here overrules the setting in Device ID. As soon as an ASIO output port is selected, ASIO will be used and the buffer filling display will turn from blue to green.
ASIO Override channel 2 ASIO right channel normal output.
See [[511].
FM 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.
Latency panel Sound card settings for Low Latency Monitoring Output.
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 General input settings.
Input Device ID Sound card to use for audio input.
Stream URL Stream address, if Input Device ID is set to Stream (via VLC)
A valid address of a stream that VLC can decode. This requires that VLC 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.
Input Buffer size Do not use this, leave at 0.
Low input level correction panel Adjusts the input level if the input level is low.
Input 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 (Link error 'Declipper', Link error 'Natural Dynamics', Noise removal) 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 with different output sound card (not ASIO) 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.
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.
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.
RC 2 Second RC circuit inversion value
See 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.
SCA Input 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. SCA panel SCA sound card settings.
Input 2 Enable the secondary (usually SCA) input.
Input 2 Device ID Sound card to use for SCA audio input.
If the input is coming from a different sound card than Input Device ID, hiccups may occur on one of the inputs.
Input 2 Separate thread Use a separate thread for SCA input.
Enable this to avoid hiccups on the main audio if there are differences in sample rate between the sound cards for Input Device ID and SCA Input.
Only enable this if it's really - if possible use the same sound card for both inputs. While this checkbox protects the main audio (Input Device ID) from hiccups, the SCA signal may still get hiccups.
Non-SCA panel Use the SCA Input for other purposes than SCA.
No SCA, add this sound card input to normal Input Combine both inputs and use the result as processing input.
This option was added for the following situation: One location where multiple signals are generated, which consist of a 'shared' part coming from one studio (which is input via one of the sound cards) and different regional programming coming from different studios (input via the other sound card).
Connecting both signals to the same sound card would cause the regional audio signals to get 'mixed', this option circumvents that problem.
Normal 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.
Normal output Enables the normal (non-FM) output.
Device ID Sound card to which the normal (non-FM) output should be sent.
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 (ASIO Override channel 1, ASIO Override channel 2), 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 Latency), set this value as low as possible without getting audio drops, and avoid running other programs that cause hiccups.
FM Output section Settings for FM output. FM panel Sound card settings for FM output.
FM output Enables FM output.
Device ID Sound card to which the normal (non-FM) output should be sent.
This can be overruled by Link error '' and Link error ''.
Volume FM output volume. Need to be calibrated for a compliant FM signal!
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 Latency), set this value as low as possible without getting audio drops, and avoid running other programs that cause hiccups.
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.
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.
RC 2 Second RC circuit inversion value
See 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.
Output & tilted output 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.
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.
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.
Synchronize to output Enables synchronization.
This increases the CPU load.
Max speed adjustment Maximum amount of speed adjustment.
See also Relative adjust. This limits the maximum amount of speed adjustment.
Turbo: Speed on start 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 speed 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.
Calibration section This is obsolete and should normally not be needed anymore. LQ Low Latency section Settings for Low Latency Monitoring Output. Latency panel Low Latency Monitoring output settings.
Low Latency output Enables the Low Latency Monitoring Output.
Device ID 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.
Buffer size Controls the delay. Set as low as possible.
