I am broadcasting a game via XFire and it uses the Windows audio device to capture any audio I receive. As I am broadcasting, other users who watch the video stream are communicating with me over Skype, and they hear themselves back within the video stream and it is entirely logical since I am broadcasting the audio I hear.
What I want to do is create another audio device within Windows and redirect (pipe) ONLY the audio input from that game and not the input reveived from Skype. I would then tell XFire to use that newly created "virtual" audio device to broadcast and therefore my partners won't hear themselves back.
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Your iPhone or iPad and your AirPlay-enabled device must be on the same Wi-Fi network before you can get automatic and suggested AirPlay connections. To get automatic and suggested AirPlay connections to your Apple TV or HomePod, make sure your Apple TV or HomePod is logged in to the same iCloud account as your iPhone or iPad.
However, now that I don't have those devices installed (somehow), I can no longer control the volume. Maxing/muting the volume only affects the host's output volume, and I have a feeling it's related to these missing devices/drivers. Help
Have been struggling with finding a way to stream audio, from a file or web, to a specific output device, not just the default one. Tried using mciSendString and while the open command does accept a device id/filename I haven't found a way to make use of it, am not even sure if this is what I am looking for or not, but considering it says ... or the filename of the device driver am guessing yes(?), but correct me if I am wrong, and this isn't a specify your output device type parameter.
If it is the correct thing then how do you enumerate the installed device drivers, have looked into the IMMDevice interface because it seamed like it could have the file names stored in the registry, but non of the output device registry keys had a driver filename type value entry, or at least I haven't found one.
So my question is, how would you go about streaming audio to a specific output device, it doesn't have to be done through mciSendString, that's just something I looked into as it's one of the most talked about function when it comes to playing audio.
To set the WaveAudio device (soundcard) used by the Multimedia Control, you must use the mciSendCommand API. The Multimedia Control does not directly provide a method to let you set the device used for playing or recording.
I have a project that includes video playback using a Media Player and a Media Sound Component for audio. It works fine when running as a stand-alone packaged game, but when I run the exact same packaged game and view it remotely via Pixel Streaming, the video plays but without its audio. Other game audio works fine also via Pixel Streaming.
Thanks for your reply.
I actually solved my problem by installing VB-Audio on the instance. This adds a virtual audio device that is always running.The windows audio device seems to be just activated when I connect to the instance via RDP.
Thanks for sharing. I was also able to get it working using VB-Audio. Also, disabling remote audio in RDP (by setting audiomode:i:1 in .rdp file) lets VB-Audio run even when you are connected via RDP.
As a suggestion, it would make for a more intuitive interface if [Streaming audio] were an item in the input pull-down list. After all, a user may want to listen to a CD at the same time as recording streaming audio in the background.
If, indeed, Audacity is paused waiting for an audio stream to start, then, I would suggest that some kind of pop-up notification should appear to let the user know that Audacity is paused, waiting for the stream to start - something like:
Likewise, if recording is paused or disabled because of some conflict related to (say) a Bluetooth audio headset - then there should be notification of that too. I have no idea what internal conflict that would be - but certainly, on my system, I cannot record audio streams if my Bluetooth headset is paired and connected to my laptop. If this happened to me, I am quite sure that others have experienced a similar frustration.
An audio endpoint device is a hardware device that either renders or captures audio. Examples include speakers, headphones, microphones, and CD players. The SAR always uses an audio rendering device. There are two ways to select the device.
The first approach is to enumerate the audio rendering devices on the system, using the IMMDeviceEnumerator interface. This interface is documented in the core audio API documentation.
Rather than enumerate devices, you can specify the audio device by its role. An audio role identifies a general category of usage. For example, the console role is defined for games and system notifications, while the multimedia role is defined for music and movies. Each role has one audio rendering device assigned to it, and the user can change these assignments. If you specify a device role, the SAR uses whatever audio device has been assigned for that role. To specify the device role, set the MF_AUDIO_RENDERER_ATTRIBUTE_ENDPOINT_ROLE attribute.
An audio session is a group of related audio streams that an application can manage collectively. The application can control the volume level and mute state of each session. Sessions are identified by GUID. To specify the audio session for the SAR, use the MF_AUDIO_RENDERER_ATTRIBUTE_SESSION_ID attribute. If you do not set this attribute, the SAR joins the default session for that process, identified by GUID_NULL.
By default, an audio session is process-specific, meaning it contains only streams from the calling process. To join a cross-process session, set the MF_AUDIO_RENDERER_ATTRIBUTE_FLAGS attribute with the value MF_AUDIO_RENDERER_ATTRIBUTE_FLAGS_CROSSPROCESS.
To control the master volume level of all the streams in the SAR's audio session, use the IMFSimpleAudioVolume interface. To control the volume of an individual stream, or to control the volume of individual channels within a stream, use the IMFAudioStreamVolume interface. Both interfaces are obtained by calling IMFGetService::GetService. You can call GetService directly on the SAR, or call it on the Media Session. Volume levels are expressed as attenuation values. For each channel, the attenuation level is the product of the master volume and the channel volume.
Although I gave up that feature , is it possible with the streaming audio file player to put it in the global rackspace and have it load a specific song based on the song selected in setlist/songpart mode?
This would indeed be a great blog article. I got the same error above and ultimately Once I get this working, I probably need to delete my audio file player in all rackspaces and then add the ability to control the SAFP from the foot switch (like I used to do with the Audio File Player)
Im trying to use my Samsung bluetooth headphones.
Was follwing the guide in the 2nd post. Please notice that this guide explain how to use osmc/raspi as an A2DP music receiver NOT as audio streaming source. So i was a bit confused. But this guide helps me to make it able to connect my headphones to my raspi.
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