Is it possible to play custom audio (*.wav file) over VOIP (SIP) without sound card being installed on SIP client machine? All my needs is to perform SIP call and play custom sound message.
You can transmit a recorded audio in the form of a WAV file over a SIP signal if you convert it to the appropriate codec first. This does not require a sound card to transmit this audio. A sound card is only required to listen to the audio. Which codec to use depends on the platform. Here is a link for converting to appropriate codecs when using Asterisk. There are a lot more if you just Google something like "audio codec conversion".
A simpler approach is to just use a platform that does this for you, like Voxeo Prophecy. This is a software only IVR solution that has a 2 port version for free. It is easy to install and program using the open standard VoiceXML. It will play back audio files recorded in a WAV file format and the telephony interface is SIP.
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My goal is to be able to write sheet music in Musescore and then have the audio output of the playback routed to Ableton Live.
I've tried using loopMIDI audio and LoopBe1 as virtual midi cables.
I have the Jack audio driver set in Ableton's audio preferences under ASIO drivers. As seen in the photo, it seems that Ableton is recognizing the virtual midi cables as an input. I have Musescore's Jack audio settings enabled. I have a midi instrument set up in Ableton. However, when I play back audio in Musescore Ableton doesn't seem to be recognizing any input.
I was trying to follow along with this tutorial. However, they seemed to omit certain details. For example, as seen in my image I was only able to route general sound/midi devices together not specific [left1,right1] to another [in1,in2]
Today I've connected my bluetooth headphones(Ausdom M08) with PC(via bluetooth dongle).
When I open Skype or Discord I hear no sound on youtube, browser and so on. It only works on Skype and Discord - bad sound, not stereo.
I checked in Sounds Options and I have Ausdom M08 Stereo and Ausdom M08 Hands-Free. First one is default device and second one is default communication device.
When I try to force Skype and Discord to use that default device for sound output I hear no sound then, too!
What I tried:
-Disabling Hands Free Telephony, but I lose microphone function then.
-Tried to uninstall drivers and install again. Still the same.
-Disabling enhancements and exclusive controls of devices.
Literally I tried everything I found on internet or that I thought it can be.
Nothing works.
So the question is: How to make my PC output Stereo Sound from my headphones and still to be able to use microphone from it?
Thanks
There are a lot of missing information here: What PC dongle are you using? Which windows version?
From what I can see about the tech specs of Ausdom M08, it supports few basic profile (HSP/HFP/A2DP/AVRCP). A2DP profile lets you hear stereo audio (Ausdom M08 Stereo). HSP/HFP lets you use the microphone to Skype, but audio is limited to 8K-16K Hz sampling rate (Ausdom M08 Hands-Free). You can't use both Bluetooth profiles at the same time.
So to answer you question: You can't have stereo audio from your headphone while having microphone input.
There are proprietary codec developed by Qualcomm called aptX, which may support microphone over A2DP. But, you'll have to make sure both transmitter and receiver supports this codec.
Can anyone please elaborate following questions?
How bluetooth stack handles audio data?
How audio commands are processed?
Did we need any service to process audio data?
Thanks in advance.
Basically, voice commands over BLE require:
some audio codec for reducing required bandwidth (ADPCM and SBC are common, OPUS is emerging),
some audio streaming method through BLE,
decoding and getting the audio stream from BLE daemon to a command processing framework.
In the android world, command processing framework is google sauce (closed) that most easily gets its audio from an ALSA device. What is left to be done is getting audio from the remote to an ALSA device.
So for audio streaming, either you:
use a custom L2CAP channel or a custom GATT service, this requires a custom android service app and/or modifications to Bluedroid to handle those, it will need a way to inject audio stream as ALSA, most probably with a "loop" audio device driver,
declare audio as custom HID reports, this way, Bluedroid injects them back to the kernel, then add a custom HID driver that processes these reports and exposes an audio device.
Audio over BLE is not standard, so all implementations do not do the actual same thing. In Nexus Player case, implementation uses HID: It streams an ADPCM audio stream, chunked in HID reports. There is a special HID driver "hid-atv-remote.c" in Android linux kernel that exposes an ALSA device in addition to input device. Bluedroid has no information about audio, all it does is forwarding HID reports from BLE to UHID.
As I understand it, streaming via bluetooth is handled via the A2DP profile. While the SBC codec is default, A2DP supports AAC, MP3, and a few other Codecs.
My question is, since spotify files are in the OGG VORBIS format (OGG Container, Vorbis Codec), what is the best way to handle streaming via Bluetooth without quality loss? Is there a specific A2DP implementation? Are folks like Jambox, etc just using the SBC implementation?
Spotify's streaming format is an implementation detail to all clients, and making the assumption that it's OGG Vorbis is not something you should do, and in some circumstances is actually a false assumption.
Since you've managed to use every single Spotify tag in your question, I don't know which platform you're developing for. However, the correct thing to do is take the PCM data the Spotify playback library gives you and use whatever playback stack your target platform gives you. On Android, iOS, Mac OS, etc the system will handle audio output devices for you, including Bluetooth streaming.
I have written an application that receives media files from a central server and plays those files according to a playlist. All works well.
A client has contacted us and wants to use our application to play some audio files as presentations in a kiosk-style application. So far, so good, our application can handle this no problems.
He has requested as a potential feature that we would have a number of headphone sockets at the front of the kiosk. Each headphone socket would play the same audio presentation in a different language.
I have come up with the idea of encoding a single audio file with the presentation in multiple languages, and each language in a different channel. We would then require a sound card that could decode each channel and output it on a different headphone socket.
Thing is, while I'm think the theory is sound, I have absolutely no idea whether this is feasible and what would be required to pull it off.
Any ideas?!
As a side-note: the application uses Media Player as the underlying component to handle the playback of audio and video. I'd appreciate any help as to the software we could use to generate the multi-channel audio stream and the hardware (USB sound card would be fine) that we could use to decode the stream.
Thanks!
You need to use multiple files not channels, its going to be way easier that way.
Instead of using Media Player use DirectShow (on .NET you have DirectShow.NET), In DirectShow you have the notation of Multiple files on the same graph.
You will be able to control to which audio device play which files, and your Play, Pause, Stop commands will be preformed on all files without you need to worry about syncing.
There are many samples on how to build media player like with DiectShow, extending them to use multiple files should be really easy.
For HW take a look at this (USB with 8 output channels)
I think with Shay's hardware you've got a complete solution:
Encode a 7.1 file with a different mono voice track on each channel.
Use the 8 channel output device in 7.1 mode, with a different headset in each port, and you've got it. Or, if you only have 6 languages, a 5.1 file would work. Many PC's have 5.1 outputs built in, you'd only need 3 splitters to break out the left and right channels from each jack.
You can do the encoding with Windows Media Encoder, or other pro audio tool.