Using the code I provided in an new empty baseplate with UseAudioApi true and EnableDefaultVoice true. For 30 seconds the memory reached over 15,924 megabytes. That is way over than Roblox server’s maximum memory of estimated 6,300 megabytes.
I’ve also noticed that this also works in Roblox server not just in Roblox studio. I have a test game here memory leak test - Roblox
Ah I had tested the code in a script with Client RunContext, and that does not leak; changing that to Server, I am seeing similar. Yikes – will look into what’s causing that.
Hey @JOSIMA33, a fix for these clicking noises should be going out in v638 – thanks for the report!
The problem occurs when creating an AudioPlayer and immediately playing it, so as a temporary workaround, creating AudioPlayers in advance, before playing them, avoids this. Even a single-frame wait is long enough:
while true do
local audioPlayer = Instance.new("AudioPlayer", workspace)
task.wait() -- waiting here avoids the click
audioPlayer.AssetId = "rbxassetid://7003103879"
audioPlayer:Play()
audioPlayer.Ended:Connect(function()
audioPlayer:Destroy()
end)
task.wait(0.1)
end
you could also pool & reuse AudioPlayers instead of calling Instance.new/Instance:Destroy,
local pool : {AudioPlayer} = {}
local function createAudioPlayer() : AudioPlayer
if #pool == 0 then
return Instance.new("AudioPlayer", workspace)
else
return table.remove(pool)
end
end
local function destroyAudioPlayer(audioPlayer : AudioPlayer)
audioPlayer.AssetId = ""
table.insert(pool, audioPlayer)
end
which only encounters the click the first few times, when the pool is empty.
@mastawba there’s a fix currently in-review for the server-side memory leak – hopefully should also be in v638 
– thank you for reporting!
I’d like to thank you for creating something this awesome.
Here’s some stuff I was able to create
Do you plan to add an audio instance that could be used for vocoding?
You’d specify a frequency band or apply a parametric eq, and it’d use the resulting audio data to manipulate another input (that could also be mixed using eqs and ducked over the background to only modulate voice)
I wonder. Will the old sounds be deprecated?
I don’t know if this has been answered before, but will audioEmitters ever work with volumetric audio?
Volumetric audio was a so much great feature. I am kinda confused why AudioEmitters don’t use it.
After working with the new audio API, I had a few ideas for features. Adding these features would make it possible to create more dynamic and immersive environments.
It would be awesome to have a property that controls the directionality of sound for the AudioListener, similar to how different microphones capture sound. The same idea could also be added to AudioEmitter but in terms of direction of sound emittance.
While AudioInteractionGroup can sometimes provide a workaround, a built-in directional sensitivity feature would be more accurate, flexible, and easier for developers to implement dynamic environments.
In a stage or presentation setting, a directional mic could prevent audience voices from being picked up, focusing only on the speaker in front of the mic. Unfortunately, listeners hear in all directions so we need a manual system.
The first technique that comes to mind is to make it so each audience user’s voice AudioEmitter has a different group than the stage mic AudioListener. This achieves a desired result but isn’t very flexible. Imagine we wanted it so that anyone can walk on stage and be picked up by the mic. They should be heard based on the direction and distance to the mic.
The current workaround would require tracking each player’s direction and position relative to the listener.
Additionally, simulating accurate distance attenuation would require a mic listener for each player/input. This is because DistanceAttenuation is only a property of AudioEmitter. Having it also for AudioListener would be useful. (Highlighted Correction: Here
With a directional property, the AudioListener can be defined as most sensitive to hearing sounds in front of it. This will ignore audience voices while still being flexible enough to automatically allow anyone to walk on stage and have their voices heard accurately.
Polar patterns are a great way to visualize what this property might allow for. Control over defining custom polar patterns would be amazing. At the very least an enum would also work for most usecases. Currently, both listener and emitters are omni-directional.
In Roblox, a visualization feature would be necessary to see how the directional sensitivity is oriented relative to the parent.
AudioListener and AudioEmitter need to be parented to anything with orientation. The “forward” direction for these audio instances are thus locked to the orientation of the parent. I propose more control over how they are oriented.
Let’s say I make a top-down experience with the camera locked to some orientation. I want spatial audio so I put an AudioListener in the character. While facing some direction a sound is heard from the right. If my character does a 180deg turn that sound will now be heard from the left which contradicts how my camera is oriented.
To workaround this I would have to make some part or attachment that has to be updated per frame to match my character position, but not rotation. The AudioListener would be parented to this extra instance.
Instead, a property could be introduced say “AlignmentMode” inspired by a similar property in constraints. If the AlignmentMode is “Parent” it will match the parent orientation like how it is currently. If the AlignmentMode is “Global” then the alignment could be locked to a specific global direction. There could also be modes for per-axis control.
If directional sensitivity is added for AudioEmitter, then this feature could also work for emitters.
Is the AudioAnalyzer client side only? Ive been playing with it and I only get an output on the client, the server always outputs 0 on both RMS and Peak:
My Setup:
Client Script Output for RMS:
Server Script Output for RMS:
I wonder. Will the old sounds be deprecated?
Hey @monkeymario1000, we will not be removing Sound, SoundGroup, or any of the SoundEffect APIs. We do think there are some sound design use-cases that just aren’t possible with those instances, but we don’t plan to delete them
will audioEmitters ever work with volumetric audio?
@BubasGaming we’ve looked at porting this over, but unfortunately it’s a little tricky now that there can be arbitrarily-many AudioListeners hearing an AudioEmitter from different perspectives.
Sounds also only support volumetric emission with Box, Cylinder, and Ball parts, and we’ve long had ambitions of making it work with arbitrary shapes (e.x. meshes).
The new API separates file playback from emission, so you might be able to accomplish something similar with multiple AudioEmitters spread around the surface of a part (e.x. offset by Attachments)
- AudioListener needs a property for controlling sensitivity of hearing sound directionally.
- AudioEmitter needs a property for controlling sensitivity of emitting sound directionally.
- AudioListener needs a DistanceAttenuation property.
- AudioListener needs properties to control the alignment of hearing spatial emitters.
- AudioEmitter needs properties to control the alignment for emittance of sound if directional sensitivity is added for emitters.
@jonbyte thanks for the feedback! We are discussing
Is the AudioAnalyzer client side only? Ive been playing with it and I only get an output on the client, the server always outputs 0 on both RMS and Peak
@jplib325 yeah, all audio processing is disabled on the server to conserve memory & cpu, which unfortunately means that AudioAnalyzer only works client-side. If you need metering information on the server, you could have each client compute volume levels for some subset of the audio (maybe their own voice or nearby sounds) and do something like what @tacheometrist shared above
right now I’m using an UnreliableRemoteEvent to share this data with the server
cc: @ReallyLongArms
I was reading through these sentences and I would like to correct what I am trying to say. The main point for this part is really wanting the addition of distance attenuation for the AudioListener. Let’s say each AudioEmitter for each player’s voice chat is set to some DistanceAttenuation curve. But then let’s also say that we want the microphone to pick up sounds in it’s own range. This is where DistanceAttenuation for AudioListener would shine.
To correct what I said, the manual approach would actually require multiple emitters (not listeners) for each player for the microphone to listen to. The DistanceAttenuation curve would be inverted for each. For example, if we wanted it so the microphone AudioListener had a max range of 100 studs then each extra player AudioEmitter DistanceAttenuation curve would be inverted such that at distance 100 the curve = 1, and at distance 0 the curve = 0. Finally, this curve would have to be updated based on the distance from the player to the microphone.
Adding DistanceAttenuation for AudioListener may still require extra AudioEmitters, but it removes the need for the weird inversion of the AudioEmitter curve. In other words, it allows prioritizing the range of the listener over the range of the emitter. However, I suppose the case of handling an AudioEmitter and AudioListener both with curves could be a thing, but I don’t think that causes any issues?
The DistanceAttenuation curve property is awesome, but for simple configuration I think it would also be cool if there was a Distance and AttenuationMode property. Modifying these would override the DistanceAttenuation curve, but it can still be edited as well (overriding the properties I mentioned to “custom”). AttenuationMode could be an Enum including Linear, Inverse, ecc. Distance would be a number.
what is the base instance for the all audio instances?
can someone help?
I have not enabled any beta features yet I still see the instances in play testing(I cant insert them in editing mode using the interface but only commands). Is it out of the beta?
is there a way to detect if someone is talking?
So I’m trying to use the audio API for a Voice Chat Radio I was programming, and I used the sample code to recreate the old voice chat RollOff, but the ListenerPosition is the workspace’s CurrentCamera’s Position.
So two players could try to speak to each other normally while being far from the CurrentCamera Position, and they can’t hear each other. How could I get around this?
what is the base instance for the all audio instances?
@VeryLiquidCold currently all the audio instances inherit from Instance. We thought about making them share a base “AudioNode” class, or inherit from “AudioProducer”/“AudioEffect”/“AudioConsumer” base classes, but since our engine only supports single-inheritance, this would close the door on some types like AudioPlayer ever inheriting from another meaningful base-class, e.x. Playable.
You can check whether a particular instance is consuming/producng audio by checking :GetConnectedWires("Input")/:GetConnectedWires("Output"); combined with pcall you could generalize this further
I have not enabled any beta features yet I still see the instances in play testing
@whatplayer5858 the studio beta only controls whether the instances are findable/insertable in the explorer; any audio instances that were already there or instantiated by scripts are still functional; we did this so that you can still publish experiences that use the new features
So I’m trying to use the audio API for a Voice Chat Radio I was programming, and I used the sample code to recreate the old voice chat RollOff, but the ListenerPosition is the workspace’s CurrentCamera’s Position.
Hey @BehzadGen, there’s an AudioEmitter:SetDistanceAttenuation API that makes implementing the equivalent rolloff much simpler; the announcement is here and
there is some sample code showing how this can be used to recreate the old voice rolloff without a per-frame/polling loop
So two players could try to speak to each other normally while being far from the CurrentCamera Position, and they can’t hear each other. How could I get around this?
You could reparent the AudioListener to your local player’s character’s head, or to a part/attachment/model that’s nearer – we spawn an AudioListener on the camera by default because we can more reliably count on it existing, but that default isn’t necessarily the best, feel free to change it for your needs!
2nd Edit
Don’t do V do what ReallyLongArms said below me their solution was so much better and worked and got rid of audio still being really faint
Fixed the issue, leaving up incase anyone else runs into the same issue:
You have to set the DeviceOutput.Player to a Player, for me it’s the Player the DeviceOutput is linked with, otherwise it’s heard by everyone.
For some reason, all Audio from AudioDeviceInput’s are heard globally (Meaning all Players can hear each other, regardless of how far they are) and I’m not sure why. I’ve tried both methods from the code samples here AudioEmitter | Documentation - Roblox Creator Hub and audio spoken is still heard globally to all Players.
Here’s my setup that builds the Audio stuff onto the player, am I missing something or is this a bug?
testing
editing
-- Services
local Players = game:GetService("Players")
local function wireConnect(Source: Instance, Destination: Instance, Name: string, Parent: Model)
-- Voice System
local Wire = Instance.new("Wire", Parent)
Wire.SourceInstance = Source
Wire.TargetInstance = Destination
Wire.Name = Name
end
Players.PlayerAdded:Connect(function(Player: Player)
-- Voice System
if Player:FindFirstChild("AudioDeviceInput") then
local DeviceInput = Player:FindFirstChild("AudioDeviceInput")
DeviceInput.Name = "DeviceInput"
DeviceInput.Player = Player
else
local DeviceInput = Instance.new("AudioDeviceInput", Player)
DeviceInput.Name = "DeviceInput"
DeviceInput.Player = Player
end
Player.CharacterAdded:Connect(function(Character: Model)
Character:MoveTo(workspace.Checkpoints[Player:GetAttribute("Checkpoint")].Base.Position)
-- Voice System
local Emitter : AudioEmitter = Instance.new("AudioEmitter", Character)
Emitter.Name = "Emitter"
local curve = {}
curve[0] = 1
curve[50] = 0.5
curve[100] = 0
Emitter:SetDistanceAttenuation(curve)
wireConnect(Player.DeviceInput, Emitter, "voiceFilterWire1", Character)
local voiceFilterWire2 = Instance.new("Wire", Character)
voiceFilterWire2.Name = "voiceFilterWire2"
local Listener = Instance.new("AudioListener", Character)
Listener.Name = "Listener"
local DeviceOutput = Instance.new("AudioDeviceOutput", Listener)
DeviceOutput.Name = "DeviceOutput"
wireConnect(Listener, DeviceOutput, "OutputWire", DeviceOutput)
local Analyzer = Instance.new("AudioAnalyzer", Character)
Analyzer.Name = "Analyzer"
wireConnect(Player.DeviceInput, Analyzer, "Wire", Analyzer)
end)
end)
This might be happening because in CharacterAdded, both an emitter and a listener are created and parented to Character – this means each voice is both emitted-from and heard-by the Character’s 3d position; the listener will hear the emitted voice as full-volume, since they are situated on top of one another.
Each listener also wires what it heard to an AudioDeviceOutput which renders it to your speakers.
Instead of creating an AudioListener & AudioDeviceOutput per-player, you could use a client-side script to just create one listener that hears things from your local perspective (e.x. workspace.CurrentCamera or Players.LocalPlayer.Character)
Edit: Ah! I see you were able to fix it by using the AudioDeviceOutput.Player property – that also works, since that will restrict which players are able to receive from a particular output.
I have actually used this and it is apparently not what I had in mind.
I was thinking that this could be used to play the same sound from all the cars in a multiple-car train, where the cars are connected together. (Copying the sound to all the cars would make it hard to change it later.)
However, I realized that what I wanted to do was practically impossible because AudioEmitter can only send one sound source.
Personally, I would have liked to have a mechanism whereby when a part containing the original sound source is connected to another part via Wire, the sound is played back exactly the same way on the other parts (even if the pitch is changed in the script, it will always be duplicated).
@ReallyLongArms Hello!
I’ve been running into an interesting dilema recently with PlaybackRegions and I was wondering if I could get some help on it.
We’re running into an issue where playing two files at once utilizing PlaybackRegions will sometimes not load properly after one another if there is decent space in between.
Are there any specific values or timeframes between when a playback region sound is called for and when it’s culled? It seems to be a really short time between and can cause sounds to choke up during gameplay.
Is there a culling difference between PlaybackRegions vs. Single playable files?
I remember that shorter files are loaded directly from memory.
When utilizing playback regions and a single file with multiple variations in it, are there any changes to caching?
If it is culling with the playback regions / audio, it seems overly aggressive at the moment.