I want to use an incoming audio stream (microphone from an external device) as the microphone input for an outbound Twilio Voice call.
The external device serves as a softphone, and does not currently support WebRTC. Instead it currently sets up 2 separate connections to a server: 1 for outgoing audio (microphone), and 1 for incoming audio. Bots connections (streams) are set up using gstreamer (gst-launch).
The server sets up a voice call and should somehow use the incoming audio stream as the microphone input for this call. I have already found the Stream instruction able to send the calls' audio back to the external device.
Can anyone point me in the right direction, maybe suggest some SDK functionality?
Related
is there any way to send Microphone audio stream to service side in real time?
I am using WCF service at the middle layer where I am converting audio to text using system.speech. It is working fine if I am sending wav file as memory stream but how it possible in a live scenario using the microphone?
Is there a way create a virtual audio output device that would make it show up in the Music app's or Spotify's output options? Alternatively, is there a way to intercept the audio stream and then force audio output to something unused (say, open headphone port)?
What I want to do is take the raw audio stream, encode/compress it via a codec, and then send over BLE (not BT Classic). Ideally my "device", or service, would show up in the output options of the music/Spotify apps
I want to develop an android app which uses push to talk or Voip to communicate between 2 mobile devices using a Wi-Fi connection without using a data or the internet.
Is it possible to develop this?
I am looking for a 'free' Wi-Fi Walkie Talkie but I don't know how to start doing it?
For a very simple, initial version, I'd do the following:
Assign a static IP address on each phone
Record the audio and packetise it into a UDP stream that you'll send to the remote IP address
Repackage again the UDP stream and play the audio.
Microphone ---> Samples ----> UDP (over WiFi) ----> Samples ----> Speaker.
There are plenty of enhancements that you can add over the time:
Device discovery
Signalling separate (SIP or custom messages via TCP or UDP to indicate when to start the audio transmission and negotiate the media channel that will be used).
Support for third party devices (SIP)
Use standard paketization using RTP and encode the audio using G711alaw/G711ulaw...
There are plenty options, but you should start by capturing the audio, and sending it over the network, even make packets of 2 seconds with the incurring delay, but that's a point to start. Then, you should lower the packet length to contain 20ms of audio to avoid delays in the transmission.
Hope this helps.
We're looking to send some serial data out from the headphone jack, but would like to still be able to play audio from the speakers. Is it possible to send output to both? If so, is it possible to send different audio to each?
Not as far as I'm aware. You can get programatic notification of when the routing has changed (i.e. when someone connects a headphone cable), but you are unable to specify which device(s) to use for output.
I want to create a voice chat software using FMOD.
Now I can receive data from microphone and play it immediately.
It's also easy to send sound data to another computer on the network.
But I don't know how to play the sent data on the other computer with FMOD.
Can anyone help me?
When you receive the incoming sound data on the destination machine you need to create a streaming buffer to play the audio. The simplest method would be to look at the userccreatedsound example. It shows how to create a custom stream buffer and use the pcmreadcallback to populate the sound with data as needed.