I'm currently developing an app for the HoloLens 2 that needs to stream audio from a desktop PC.
The idea is to send control information (position, orientation, etc.) to a Cycling'74 Max/Msp application running on a Windows 10 computer to process audio for 3D audio playback. I now need to somehow stream the resulting sound to the Unity app running on the HoloLens. Both devices run on the same network.
At the moment I've achieved something using mrtk webrtc for unity in combination with a virtual cable as input. My issue is that this seems to be optimized for microphone use as it applies some options like noise reduction and smaller bandwidth. I can't find a way to set the options for webrtc to stream what I need (music) with better quality.
Does anyone know how to change that on mrtk webrtc or has a better solution for the audio streaming to the hololens?
WebRTC project for Mixed Reality is deprecated and it is designed for real-time communication. If your requirement is media consumption, you need other workaround solutions.
For dedicated media streaming, you can set up a DLNA server on your PC for media access.
You may also set up Samba or NFS on your PC if you need to access files in other formats.
Related
I am looking for a way to record microphone input from a specific channel.
For example, I want to record separately left/right of an M-Track audio interface or SingStar wireless microphones.
Microphone.Start seems limited in this regard.
Further, I found this thread, which says
Yes, you can assume the microphone at runtime will always be 1 channel.
My questions:
Any workaround to get this working?
It there maybe an open source lib or low level microphone API in Unity?
Is it really not possible to record different microphone channels into different AudioClips in Unity?
I am looking for a solution at least on desktop platform, i.e., Windows, MacOS, Linux.
Bonus question: Is recording specific microphone channels working in Unreal Engine?
I'm a bit lost looking through all the various Agora.io modules (and not sure what it means that only some of them have Unity-specific downloads).
I want to make a Unity app where two remote phones exchange data as follows:
Streaming voice in both directions
Streaming video in one direction (recorded from device camera)
Streaming a small amount of continuously-changing custom data in the other direction (specifically, a position + orientation in a virtual world; probably encoded as 7 floats)
The custom data needs to have low latency but does not need reliability (it's fine if some updates get lost; app only cares about the most recent update). Updates basically every frame.
Ideally I want to support both Android and iOS.
I started looking at Agora video (successfully built a test project) and it seems like it will cover the voice and video, but I'm struggling to find a good way to send the custom data (position + orientation). It's probably theoretically possible to encode it as a custom video feed but that sounds complex and inefficient. Is there some out-of-band signalling mechanism I could use to send some extra data alongside/instead of a video?
Agora real-time messaging sounds like it would probably work for this, but I can't seem to find any info about integrating it with Unity (either on Agora's web site or in a general web search). Can I roll this in somehow?
Agora interactive gaming could maybe also be relevant? The overview doesn't seem real clear about how it's different from regular Agora video. I suspect it's overkill but that might be fine if there isn't a large performance cost.
Could anyone point me in the right direction?
I would also consider alternatives to Agora if there's a better plugin for implementing this feature set in Unity.
Agora's Video SDK for Unity supports exporting projects to Android, iOS, MacOS, and Windows (non-UWP).
Regarding your data streaming needs, Agora's RTM SDK is in the process of being ported to work within Unity. At the moment the best way to send data using the Agora SDK is to use CreateDataStream to leverage Agora's ability to open a data stream that is sent along with the frames. Data stream messages are limited to 1kb per frame and 30kb/s so I would be cautious about running it on every frame if you are using a frame-rate above 30fps.
I can realize sending data from HoloLens (using Unity coding by C#) to PC (coding by C#) by socket communication. But how to sending video steaming in real-time (the video starts to be recorded when I open the application in HoloLens) from HoloLens to PC by my original socket frame. In my view, maybe I should add some sentences to recognize the HoloLens camera, record video and encode the video to data, then transmit the data by my previous socket. Is it right and how to realize it?
By the way, I hope that the PC can receive the video by python so that I can process the video in the following steps.
To send video steaming in real-time between HoloLens and PC client, WebRTC should can meets your needs. Please check out this MixedReality-WebRTC project, it can help you to integrate peer-to-peer real-time audio and video communication into your application. It also implements local video capture you need and encapsulation it as a Unity3D component for rapid prototyping and integration.
You can read its official documentation via this link: MixedReality-WebRTC 1.0.0 documentation.
Moreover, this project can be used in desktop applications or even other non-mixed reality applications, which can save your development costs.
I wanted to know if Unity3D supports multi-channel audio input. I am trying to use an audio interface to input and process stereo audio in Unity.
Thanks!
As far as I know, Unity supports multiple simultaneous input devices (see Microphone), but assumes only one channel per device. This a a bit of a limitation for multi channel device users.
I worked around it for my project by routing each input channel from my sound card to a virtual device, so that each device that Unity interacts with contains only a single input channel. You can do this using Jack or VoiceMeeter, for example.
Developed an internet radio streamer using jPlayer which utilizes the html5 audio tags with jQuery and has a flash fall back for unsupported browsers. Upon testing the player on the iPhone (iOS 5.0.1), we ran into a very peculiar issue.
When the iPhone is connected to WiFi, it streams perfectly using the HE-AAC V2 stream # 64kbps 44.1kHz (the preferred codec for apple products). However, when the iPhone is connected to the 3G mobile network, it "stutters" or stops streaming for 1-2 secs every 1-2 minutes (does not stop streaming completely). The troubling thing is when the iPhone is forced to use a separate MP3 stream at the same bit rate, it does not have this issue and works very well on 3G.
UPDATE 5
We recently acquired a 3G/4G Sprint mobile hotspot device and tested this issue with the device. When the iPhone is connected to the mobile hotspot, it shows as being connected to a wifi device and the issue does not render even tho the actual connection is via 3G/4G. This might point back to the issue being with the iPhone not handling HE-AAC via HTTP Live Streaming and when directly connected to the mobile network.
UPDATE 4
Updated the iPhone to iOS 5.1 yet the issue persists.
UPDATE 3
Read here on SO various issues of script not rendering correctly when connected to mobile networks. The finger seems to point to the mobile network carriers that may be inserting Proxy to serve webpages, e.g. for downsizing images. Also it might inject some JavaScript pages. The test page can be found HERE Note: this page is using HE-AAC so it will only work on iPhone...
UPDATE
According to Apple's HTTP Live Streaming doc for iOS devices, that "Audio-only content can be either MPEG-2 transport or MPEG elementary audio streams, either in AAC format with ADTS headers or in MP3 format." Our music server is using OddcastV3 encoder to send out three streams (MP3, HE-AAC V2, and Oggvorbis) to the icecastV2 server. Not sure if the encoder is inserting ADTS headers for the HE-AAC V2 stream. Is there a way to check for this?
Comming from a radio planning point of view - here are my two cents:
What you are describing sounds like bandwidth shaping - which is both a common and often neccesary design of radio networks (like 3G networks). In most 3G operators I worked at you would typically optimize your network to give high-speed burst (think downloading an image, sending one email or fetching one HTML page) - over "long-running" high bandwidth services.
This is due to the simple fact that this is what most users want/need.
This shaping can on a typical 3GPP (GSM 3G) network result in that you will first get a RAB (radio access bearer) supporting 384kbit and is then downgraded as long as your your device accepts it.
This means that typicall you will get switched from 384 -> 256 -> 128, then 64kbit where maybe your device starts recieving data to slowly, then the network upgrades it and again downgrade it after a while.
So why is not then the MP3 file stuttering? my guess is that the total kbit rate might differ - so you are fine in the 64kbit RAB. This is a common phenomena.
We have managed to get the exact same thing working. 64kbit AAC-v2 on mobile devices. We are streaming files and not a steady stream, I think Magnus is right when he explains how the network prioritized traffic to bursts, in our case that means we have large parts of the file right away and the player can continue to play until he next burst comes in. In your case that means the stream pauses until the next burst comes.
Either if you can switch to larger chunks in your streaming (larger buffer) or stream whole files instead?
We had a very strange phenomenon with iOS, we had to rename all files from .m4a to .aac to be able to get them streaming on iOS. If we didn't rename them iOS wouldnt play them.
Good luck.