Audio Repeater App
Finlay Mange
Audio Repeater allows you to transfer (copy) sounds from an input (recording) audio endpoint/device to an output (playback) one. Audio transfer can be performed in any PCM audio format supported by Windows.
MME version works through Windows Core Audio Subsystem that shares every audio endpoint among any number of audio applications. You can run several Audio Repeater instances to copy sounds from a single recording endpoint, or to mix sounds to a single playback endpoint, or combine copy/mix operations. However, due to multi-level buffering, this version adds a noticeable (dozens or even hundreds of milliseconds) latency.
KS version communicates directly with audio driver, bypassing all additional Windows audio processing. Therefore, this version allows you to achieve best performance and lowest latency, but most audio drivers support only a single client application. It means that if the endpoint is used by Windows Audio Subsystem, it cannot be accessed by the Repeater, and vice versa.
I left my pc untouched for about 2 days, when I came back this error popped up whenever the autostart executed (I used the guide here in the wiki to setup Hesuvi). I tested around and found out it fixes itself after I restart the audio service using the feature in Hesuvi, but this is bothersome to me as it wasn't like this before I left my pc. Did anybody else go through the same experience and found a fix? Thanks in advance.
I have tried to connect two of my JBL Flip 6 devices simultaneously via Audio Repeater, but it does not connect. I have tried connecting by changing the options several times but still it does not work.
In first two attempts, you tried to use sampling rate 48000 while JBL only supports 44100. The error message explains this, error "there is no match..." means that the driver didn't find the requested format in its list of supported formats.
In third attempt, the requested format is supported, but the driver was unable to create a pin instance for some reason ("a device is not functioning"). For example, it might occur if some application(s) and/or system service(s) use the endpoint associated with this pin. Try to close all audio-related applications before clicking Start.
The Sd-10 is a complete stereo audio repeater. It can be used in Store-Casting, Music-On-Hold, Museum, Haunted Attraction, Safety, Industrial or Entertainment applications. Anywhere you need a solid state, high quality audio system that will play for years. The Sd-10 can be dropped right into an audio system in place of a CD player.
I'm implementing a program that reads an audio stream from an input device and sends it to an output device using NAudio. To do that, I get the data from the input stream using WaveIn and its DataAvailable event. To write the data, I am about to use WaveOut, which also means I need to use some implementation of IWaveProvider. Implementing that for my needs seems quite straightforward using a queue. Except for one thing: what should happen when the queue is empty and the Read() method is called? I didn't find anything about this in the documentation. Options I think I have:
I was thinking about using option 1 or 2, but then I found BufferedWaveProvider (and WaveInProvider, that uses BufferedWaveProvider internally) that uses option 3. From that, I inferred that option 3 is the preferred one.
The Read method is called when the soundcard is in need of more data to play. Blocking is not usually a good option because you might be on a thread from within the driver itself (as with ASIO or WaveOut with function callbacks), or you might be on the GUI thread (as with WaveOut with windows message callbacks).
Audio Repeater allows you to transfer audio samples from one device to another with DSP processing support. For an instance, let's say you have Bluetooth headset & you want to listen to the audio output of your PC through the headset. Then you can use this tool to accomplish that task with minimum amount of latency.
Please change Buffer Count & Buffer Size if you hear gaps in the audio stream. When changing those values make sure to put right values according to the tutorial. These values are heavily depending on the soundcard & the performance of your PC.
Audio Repeater uses first DSP module when using it with Winamp DSP plug-in. This feature will allow you to use this tool as a real-time audio effect processor. Also, you can use Virtual Audio Cables if you want to apply DSP effects to the entire system. (Tested with Virtual Audio Cable v4.09)
Please note that the unregistered version has a periodic silence. To remove it, you need to register this software. The registration fee is only $35. (Previously registered users can use their old serial number)
You can use the soft audio adjustments through the SSH interface. Once you login via ssh, you can open the alsamixer utility to adjust the audio, but the adjustment levels are usually very rough (20%) so you are better off using a potentiometer in the audio path for fine adjustments on both the input and output audio.
Hello Milan,
What hardware are you using for your audio interface? If you are saying that your audio is dull, there could be a number of things to look at. The SVXLink core supports pre-emphasis and de-emphasis in software, but ORP currently sets this at a fixed setting (No UI control). Some interfaces like the ICS Controllers interfaces have hardware selectable pre-emphasis and de-emphasis. It may also depend on how you are interfacing with your radio as well or settings withing the radio. I know for instances on the old Motorolas you have different audio pins for flat vs modified audio and there are also some settings in the Motorola CPS software. So there are a couple places where things can get messed up.
As for the EchoLink. The SVXLink module offers no settings for adjusting the audio levels. You may be best to use Echolink as your standard and adjust your repeater input audio to match that. Then it should output at around the same level. You may also want to make sure that you are using multiple EchoLink sources to confirm your claim. I would also use that source and try connecting to the Echolink test server (9999) to see how the replayed audio compares with the level of the prompts. Getting everything setup correctly can be a fine balancing act since there are a number of controls in the audio path that can affect one another.
about the echolink problem, its sad to hear there is no gain setting for echolink.
RX audio is perfect, no audio hiss nice and hot but not over driven. ofcorse i can put a attenuator on the input but than i have to build an amp stage asswel to makeup for the attenuator (since the soundcard is maxed allready)
Radio Wouxun KG-935G
There is a repeater nearby that I used to be able to hear, however as of late I can "see" transmission on the receive signal gauge on my radio but I can't hear anything. The incoming signal is very strong (according to the scale), but there is not sound (no static nothing). Squelch is set to almost the lowest setting. Volume on the radio is all the way and I've made sure the repeater tone is correct.
My main question is, what setting in the svxlink.php file associates with the PCM setting in Alsamixer? If I can find that I can set the proper audio level and be done with getting this thing tweaked.
By default the settings I believe are around 70% in alsamixer and this should be plenty loud. Beyond this it sounds like potentially a deaf TX radio, is the input levels on your tx rig programmable? We have encountered this a few times now where the input levels are incorrect inside the radio.
When I try setting the audio in Alsamixer, I have to set the PCM level to about 86 to get the audio loud enough. But, the settings only stay until I either reboot/power cycle the Pi. When it comes back up, the settings revert to normal.
But to reiterate, the Alsamixer settings revert back to the default levels when I reboot. Matter of fact, every time I reboot, I have to go into the interface tab and reselect the pre-setting for the Pi 1x controller, before it will receive or transmit.
If this continues to be an issue, I would suggest writing down your settings and reimaging the sdcard or grabbing a seconds SD card (Be sure to use name brand high speed ones) and using it to experiment with. On my setup that acted this way, this reimaging of the card immediately cleared up the problem after going several rounds with the tech support for the product thinking it was a driver issue.
In the CPS utility, there is a gain (-dB as I recall) setting for the external mic connector on the back of the radio, as I recall we had to turn this up almost to maximum with the sound set at 60% (the previous default in the ICS images). We then found out through trial and error with the OR image that on the CDM radios we had, that the 70% defaults should be plenty loud. On the units I touched for a customer, this mic setting was turned way down, not sure if that is default or someone had messed with it while experimenting.
First, seems like the suggestion to turn up the mic gain worked well. Its now plenty loud (voice and synthesized voice). Now I just need to adjust the POTS on the board because the incoming voice audio is TOO loud now! (Better than it was though).
For the second one, this is an issue we recently identified as a race condition in the system booting up sequence. I will need to chase down the file path, but its a simple file edit once I find the details of which file to edit. It always takes me a while to find that pesky file.
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