I want to simulate channel distortion/noise such as telephony, VOIP and etc. on some previously recorded audio files. Can someone guide me by recommending some tools in this area?
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Plivo has mentioned on their Voice SDK features that Plivo supports Dual-channel call recording
and Recordings are dual-channel by default and stored encrypted. However I do not see where to get two different audio urls from the call recording.
Does anyone has idea how to do it?
Plivo makes mono recordings of conference calls and stereo recordings of regular calls. Having said that, for dual-channel recording(stereo) Plivo provides the audio of both the participants in separate channels.
You can listen to the audio of:
participant1 channel (the left speaker when played with stereo speakers).
participant2 channel (the right speaker when played with stereo speakers).
Hope this is helpful, please refer to our Record Calls guide for more details about the recording feature. Thanks!
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]
I am trying to build an open-source in-ear monitoring system. I have created the UI and was wondering how I would get the channels that are on an audio mixing console so that I can edit the channels and stream them to each musician. Is there a certain protocol that all the mixers use? You can find the project at https://gitlab.com/openstagemix. We would love to have contributors.
I can't really test whether this is the correct answer as I am trapped in my house during the coronavirus time. But, all mixers use something called OSC which is a protocol between mixers, synthesizers, etc. to computers. You can find more information here http://opensoundcontrol.org/introduction-osc.
Update:
It's neither! I am going to use the AES67 standard to receive information from my mixer and with that process the audio. This is because my mixer is ethernet capable.
Listening for an Audio Signal on a line in or via microphone. When detected, record audio file.
When finished send audio mp3 file to email source.
Anyone know of a solution out there in the marketplace or can code this please advise. I will pay.
Thank you
The behavior is called Sound detection or Microphone monitoring software. but Stackoverflow is not a place to recommend specific software.
We recently built a demo application utilizing Kurento Media Server to record applicant video interview, but the audio quality is not well , some audio is not recognizable and some of it had high pitch noise. We've been test it on several models of PC or Mac, so this should not be device problem.
We've been using RecorderEndpoint with media profile MediaProfileSpecType.WEBM ,and all other setting remain as default.
To fix this problem, we tried:
We upgrade to Kurento 6.2.1 which use Opus as the audio encoder.
Try to using setMaxOuputBitrate of the recorder, we don't see it has improvements or I don't know which bit rate range can be used.
Change SDPOffer to setup a high bit rate audio for Opus which we don't know where to modify
None of it is working so far, so please tell us where to look.
Thanks.
Please check with this recording tutorial. The audio should be fine. Just make sure you are only sending audio, and not video. That should help.
If the audio is not being recorded correctly, I would try and hear what's coming out of your box through your browser. Try and run the hello-world tutorial, with a pair of headphones connected to your box so you don't have echoes.
About #2, if you want to raise the bitrate exchanged between the webrtc endpoint and the recorder, you need to invoke the setOutputBitrate command on the webrtc endpoint.